/* * Assorted DPCM codecs * Copyright (c) 2003 The FFmpeg project * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * Assorted DPCM (differential pulse code modulation) audio codecs * by Mike Melanson (melanson@pcisys.net) * Xan DPCM decoder by Mario Brito (mbrito@student.dei.uc.pt) * for more information on the specific data formats, visit: * http://www.pcisys.net/~melanson/codecs/simpleaudio.html * SOL DPCMs implemented by Konstantin Shishkov * * Note about using the Xan DPCM decoder: Xan DPCM is used in AVI files * found in the Wing Commander IV computer game. These AVI files contain * WAVEFORMAT headers which report the audio format as 0x01: raw PCM. * Clearly incorrect. To detect Xan DPCM, you will probably have to * special-case your AVI demuxer to use Xan DPCM if the file uses 'Xxan' * (Xan video) for its video codec. Alternately, such AVI files also contain * the fourcc 'Axan' in the 'auds' chunk of the AVI header. */ #include "libavutil/intreadwrite.h" #include "avcodec.h" #include "bytestream.h" #include "codec_internal.h" #include "internal.h" #include "mathops.h" typedef struct DPCMContext { int16_t array[256]; int sample[2]; ///< previous sample (for SOL_DPCM) const int8_t *sol_table; ///< delta table for SOL_DPCM } DPCMContext; static const int32_t derf_steps[96] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767, }; static const int16_t interplay_delta_table[] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 38, 39, 40, 41, 42, 43, 47, 51, 56, 61, 66, 72, 79, 86, 94, 102, 112, 122, 133, 145, 158, 173, 189, 206, 225, 245, 267, 292, 318, 348, 379, 414, 452, 493, 538, 587, 640, 699, 763, 832, 908, 991, 1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993, 2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008, 4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059, 8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206, 17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589, -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1, 1, 1, 5481, 10503, 15105, 19322, 23186, 26728, 29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298, -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597, -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772, -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373, -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180, -1081, -991, -908, -832, -763, -699, -640, -587, -538, -493, -452, -414, -379, -348, -318, -292, -267, -245, -225, -206, -189, -173, -158, -145, -133, -122, -112, -102, -94, -86, -79, -72, -66, -61, -56, -51, -47, -43, -42, -41, -40, -39, -38, -37, -36, -35, -34, -33, -32, -31, -30, -29, -28, -27, -26, -25, -24, -23, -22, -21, -20, -19, -18, -17, -16, -15, -14, -13, -12, -11, -10, -9, -8, -7, -6, -5, -4, -3, -2, -1 }; static const int8_t sol_table_old[16] = { 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0 }; static const int8_t sol_table_new[16] = { 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15 }; static const int16_t sol_table_16[128] = { 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080, 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120, 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0, 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230, 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280, 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0, 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320, 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370, 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0, 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480, 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700, 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00, 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000 }; static av_cold int dpcm_decode_init(AVCodecContext *avctx) { DPCMContext *s = avctx->priv_data; int i; if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) { av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n"); return AVERROR(EINVAL); } s->sample[0] = s->sample[1] = 0; switch(avctx->codec->id) { case AV_CODEC_ID_ROQ_DPCM: /* initialize square table */ for (i = 0; i < 128; i++) { int16_t square = i * i; s->array[i ] = square; s->array[i + 128] = -square; } break; case AV_CODEC_ID_SOL_DPCM: switch(avctx->codec_tag){ case 1: s->sol_table = sol_table_old; s->sample[0] = s->sample[1] = 0x80; break; case 2: s->sol_table = sol_table_new; s->sample[0] = s->sample[1] = 0x80; break; case 3: break; default: av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n"); return -1; } break; case AV_CODEC_ID_SDX2_DPCM: for (i = -128; i < 128; i++) { int16_t square = i * i * 2; s->array[i+128] = i < 0 ? -square: square; } break; case AV_CODEC_ID_GREMLIN_DPCM: { int delta = 0; int code = 64; int step = 45; s->array[0] = 0; for (i = 0; i < 127; i++) { delta += (code >> 5); code += step; step += 2; s->array[i*2 + 1] = delta; s->array[i*2 + 2] = -delta; } s->array[255] = delta + (code >> 5); } break; default: break; } if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3) avctx->sample_fmt = AV_SAMPLE_FMT_U8; else avctx->sample_fmt = AV_SAMPLE_FMT_S16; return 0; } static int dpcm_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt) { int buf_size = avpkt->size; DPCMContext *s = avctx->priv_data; int out = 0, ret; int predictor[2]; int ch = 0; int stereo = avctx->ch_layout.nb_channels - 1; int16_t *output_samples, *samples_end; GetByteContext gb; if (stereo && (buf_size & 1)) buf_size--; bytestream2_init(&gb, avpkt->data, buf_size); /* calculate output size */ switch(avctx->codec->id) { case AV_CODEC_ID_ROQ_DPCM: out = buf_size - 8; break; case AV_CODEC_ID_INTERPLAY_DPCM: out = buf_size - 6 - avctx->ch_layout.nb_channels; break; case AV_CODEC_ID_XAN_DPCM: out = buf_size - 2 * avctx->ch_layout.nb_channels; break; case AV_CODEC_ID_SOL_DPCM: if (avctx->codec_tag != 3) out = buf_size * 2; else out = buf_size; break; case AV_CODEC_ID_DERF_DPCM: case AV_CODEC_ID_GREMLIN_DPCM: case AV_CODEC_ID_SDX2_DPCM: out = buf_size; break; } if (out <= 0) { av_log(avctx, AV_LOG_ERROR, "packet is too small\n"); return AVERROR(EINVAL); } if (out % avctx->ch_layout.nb_channels) { av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n"); } /* get output buffer */ frame->nb_samples = (out + avctx->ch_layout.nb_channels - 1) / avctx->ch_layout.nb_channels; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; output_samples = (int16_t *)frame->data[0]; samples_end = output_samples + out; switch(avctx->codec->id) { case AV_CODEC_ID_ROQ_DPCM: bytestream2_skipu(&gb, 6); if (stereo) { predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16); predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16); } else { predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16); } /* decode the samples */ while (output_samples < samples_end) { predictor[ch] += s->array[bytestream2_get_byteu(&gb)]; predictor[ch] = av_clip_int16(predictor[ch]); *output_samples++ = predictor[ch]; /* toggle channel */ ch ^= stereo; } break; case AV_CODEC_ID_INTERPLAY_DPCM: bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */ for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) { predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16); *output_samples++ = predictor[ch]; } ch = 0; while (output_samples < samples_end) { predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)]; predictor[ch] = av_clip_int16(predictor[ch]); *output_samples++ = predictor[ch]; /* toggle channel */ ch ^= stereo; } break; case AV_CODEC_ID_XAN_DPCM: { int shift[2] = { 4, 4 }; for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16); ch = 0; while (output_samples < samples_end) { int diff = bytestream2_get_byteu(&gb); int n = diff & 3; if (n == 3) shift[ch]++; else shift[ch] -= (2 * n); diff = sign_extend((diff &~ 3) << 8, 16); /* saturate the shifter to 0..31 */ shift[ch] = av_clip_uintp2(shift[ch], 5); diff >>= shift[ch]; predictor[ch] += diff; predictor[ch] = av_clip_int16(predictor[ch]); *output_samples++ = predictor[ch]; /* toggle channel */ ch ^= stereo; } break; } case AV_CODEC_ID_SOL_DPCM: if (avctx->codec_tag != 3) { uint8_t *output_samples_u8 = frame->data[0], *samples_end_u8 = output_samples_u8 + out; while (output_samples_u8 < samples_end_u8) { int n = bytestream2_get_byteu(&gb); s->sample[0] += s->sol_table[n >> 4]; s->sample[0] = av_clip_uint8(s->sample[0]); *output_samples_u8++ = s->sample[0]; s->sample[stereo] += s->sol_table[n & 0x0F]; s->sample[stereo] = av_clip_uint8(s->sample[stereo]); *output_samples_u8++ = s->sample[stereo]; } } else { while (output_samples < samples_end) { int n = bytestream2_get_byteu(&gb); if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F]; else s->sample[ch] += sol_table_16[n & 0x7F]; s->sample[ch] = av_clip_int16(s->sample[ch]); *output_samples++ = s->sample[ch]; /* toggle channel */ ch ^= stereo; } } break; case AV_CODEC_ID_SDX2_DPCM: while (output_samples < samples_end) { int8_t n = bytestream2_get_byteu(&gb); if (!(n & 1)) s->sample[ch] = 0; s->sample[ch] += s->array[n + 128]; s->sample[ch] = av_clip_int16(s->sample[ch]); *output_samples++ = s->sample[ch]; ch ^= stereo; } break; case AV_CODEC_ID_GREMLIN_DPCM: { int idx = 0; while (output_samples < samples_end) { uint8_t n = bytestream2_get_byteu(&gb); *output_samples++ = s->sample[idx] += (unsigned)s->array[n]; idx ^= 1; } } break; case AV_CODEC_ID_DERF_DPCM: { int idx = 0; while (output_samples < samples_end) { uint8_t n = bytestream2_get_byteu(&gb); int index = FFMIN(n & 0x7f, 95); s->sample[idx] += (n & 0x80 ? -1: 1) * derf_steps[index]; s->sample[idx] = av_clip_int16(s->sample[idx]); *output_samples++ = s->sample[idx]; idx ^= stereo; } } break; } *got_frame_ptr = 1; return avpkt->size; } #define DPCM_DECODER(id_, name_, long_name_) \ const FFCodec ff_ ## name_ ## _decoder = { \ .p.name = #name_, \ .p.long_name = NULL_IF_CONFIG_SMALL(long_name_), \ .p.type = AVMEDIA_TYPE_AUDIO, \ .p.id = id_, \ .p.capabilities = AV_CODEC_CAP_DR1, \ .priv_data_size = sizeof(DPCMContext), \ .init = dpcm_decode_init, \ .decode = dpcm_decode_frame, \ .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \ } DPCM_DECODER(AV_CODEC_ID_DERF_DPCM, derf_dpcm, "DPCM Xilam DERF"); DPCM_DECODER(AV_CODEC_ID_GREMLIN_DPCM, gremlin_dpcm, "DPCM Gremlin"); DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay"); DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ"); DPCM_DECODER(AV_CODEC_ID_SDX2_DPCM, sdx2_dpcm, "DPCM Squareroot-Delta-Exact"); DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol"); DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");