347 lines
12 KiB
C
347 lines
12 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 file is part of Libav.
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*
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* Libav 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 2.1 of the License, or (at your option) any later version.
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*
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* Libav 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 Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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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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* Xan DPCM decoder by Mario Brito (mbrito@student.dei.uc.pt)
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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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* SOL DPCMs implemented by Konstantin Shishkov
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*
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* Note about using the Xan DPCM decoder: Xan DPCM is used in AVI files
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* found in the Wing Commander IV computer game. These AVI files contain
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* WAVEFORMAT headers which report the audio format as 0x01: raw PCM.
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* Clearly incorrect. To detect Xan DPCM, you will probably have to
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* special-case your AVI demuxer to use Xan DPCM if the file uses 'Xxan'
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* (Xan video) for its video codec. Alternately, such AVI files also contain
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* the fourcc 'Axan' in the 'auds' chunk of the AVI header.
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*/
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#include "internal.h"
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#include "mathops.h"
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typedef struct DPCMContext {
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AVFrame frame;
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int16_t roq_square_array[256];
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int sample[2]; ///< previous sample (for SOL_DPCM)
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const int8_t *sol_table; ///< delta table for SOL_DPCM
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} DPCMContext;
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static const int16_t 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 const int8_t sol_table_old[16] = {
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0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
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-0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
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};
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static const int8_t sol_table_new[16] = {
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0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
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0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
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};
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static const int16_t sol_table_16[128] = {
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0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
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0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
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0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
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0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
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0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
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0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
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0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
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0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
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0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
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0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
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0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
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0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
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0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
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};
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static av_cold 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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if (avctx->channels < 1 || avctx->channels > 2) {
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av_log(avctx, AV_LOG_INFO, "invalid number of channels\n");
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return AVERROR(EINVAL);
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}
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s->sample[0] = s->sample[1] = 0;
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switch(avctx->codec->id) {
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case AV_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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int16_t 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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case AV_CODEC_ID_SOL_DPCM:
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switch(avctx->codec_tag){
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case 1:
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s->sol_table = sol_table_old;
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s->sample[0] = s->sample[1] = 0x80;
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break;
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case 2:
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s->sol_table = sol_table_new;
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s->sample[0] = s->sample[1] = 0x80;
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break;
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case 3:
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break;
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default:
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av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
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return -1;
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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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if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
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avctx->sample_fmt = AV_SAMPLE_FMT_U8;
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else
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avcodec_get_frame_defaults(&s->frame);
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avctx->coded_frame = &s->frame;
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return 0;
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}
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static int dpcm_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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int buf_size = avpkt->size;
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DPCMContext *s = avctx->priv_data;
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int out = 0, ret;
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int predictor[2];
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int ch = 0;
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int stereo = avctx->channels - 1;
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int16_t *output_samples, *samples_end;
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GetByteContext gb;
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if (stereo && (buf_size & 1))
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buf_size--;
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bytestream2_init(&gb, avpkt->data, buf_size);
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/* calculate output size */
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switch(avctx->codec->id) {
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case AV_CODEC_ID_ROQ_DPCM:
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out = buf_size - 8;
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break;
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case AV_CODEC_ID_INTERPLAY_DPCM:
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out = buf_size - 6 - avctx->channels;
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break;
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case AV_CODEC_ID_XAN_DPCM:
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out = buf_size - 2 * avctx->channels;
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break;
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case AV_CODEC_ID_SOL_DPCM:
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if (avctx->codec_tag != 3)
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out = buf_size * 2;
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else
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out = buf_size;
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break;
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}
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if (out <= 0) {
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av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
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return AVERROR(EINVAL);
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}
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/* get output buffer */
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s->frame.nb_samples = out / avctx->channels;
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if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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output_samples = (int16_t *)s->frame.data[0];
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samples_end = output_samples + out;
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switch(avctx->codec->id) {
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case AV_CODEC_ID_ROQ_DPCM:
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bytestream2_skipu(&gb, 6);
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if (stereo) {
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predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
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predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
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} else {
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predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
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}
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/* decode the samples */
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while (output_samples < samples_end) {
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predictor[ch] += s->roq_square_array[bytestream2_get_byteu(&gb)];
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predictor[ch] = av_clip_int16(predictor[ch]);
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*output_samples++ = predictor[ch];
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/* toggle channel */
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ch ^= stereo;
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}
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break;
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case AV_CODEC_ID_INTERPLAY_DPCM:
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bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
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for (ch = 0; ch < avctx->channels; ch++) {
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predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
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*output_samples++ = predictor[ch];
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}
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ch = 0;
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while (output_samples < samples_end) {
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predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)];
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predictor[ch] = av_clip_int16(predictor[ch]);
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*output_samples++ = predictor[ch];
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/* toggle channel */
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ch ^= stereo;
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}
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break;
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case AV_CODEC_ID_XAN_DPCM:
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{
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int shift[2] = { 4, 4 };
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for (ch = 0; ch < avctx->channels; ch++)
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predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
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ch = 0;
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while (output_samples < samples_end) {
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int diff = bytestream2_get_byteu(&gb);
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int n = diff & 3;
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if (n == 3)
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shift[ch]++;
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else
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shift[ch] -= (2 * n);
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diff = sign_extend((diff &~ 3) << 8, 16);
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/* saturate the shifter to a lower limit of 0 */
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if (shift[ch] < 0)
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shift[ch] = 0;
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diff >>= shift[ch];
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predictor[ch] += diff;
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predictor[ch] = av_clip_int16(predictor[ch]);
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*output_samples++ = predictor[ch];
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/* toggle channel */
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ch ^= stereo;
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}
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break;
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}
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case AV_CODEC_ID_SOL_DPCM:
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if (avctx->codec_tag != 3) {
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uint8_t *output_samples_u8 = s->frame.data[0],
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*samples_end_u8 = output_samples_u8 + out;
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while (output_samples_u8 < samples_end_u8) {
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int n = bytestream2_get_byteu(&gb);
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s->sample[0] += s->sol_table[n >> 4];
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s->sample[0] = av_clip_uint8(s->sample[0]);
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*output_samples_u8++ = s->sample[0];
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s->sample[stereo] += s->sol_table[n & 0x0F];
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s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
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*output_samples_u8++ = s->sample[stereo];
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}
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} else {
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while (output_samples < samples_end) {
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int n = bytestream2_get_byteu(&gb);
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if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
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else s->sample[ch] += sol_table_16[n & 0x7F];
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s->sample[ch] = av_clip_int16(s->sample[ch]);
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*output_samples++ = s->sample[ch];
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/* toggle channel */
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ch ^= stereo;
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}
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}
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break;
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}
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*got_frame_ptr = 1;
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*(AVFrame *)data = s->frame;
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return avpkt->size;
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}
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#define DPCM_DECODER(id_, name_, long_name_) \
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AVCodec ff_ ## name_ ## _decoder = { \
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.name = #name_, \
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.type = AVMEDIA_TYPE_AUDIO, \
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.id = id_, \
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.priv_data_size = sizeof(DPCMContext), \
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.init = dpcm_decode_init, \
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.decode = dpcm_decode_frame, \
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.capabilities = CODEC_CAP_DR1, \
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.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
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}
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DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
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DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
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DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
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DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
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