1/* 2 * Direct Stream Digital (DSD) decoder 3 * based on BSD licensed dsd2pcm by Sebastian Gesemann 4 * Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved. 5 * Copyright (c) 2014 Peter Ross 6 * 7 * This file is part of FFmpeg. 8 * 9 * FFmpeg is free software; you can redistribute it and/or 10 * modify it under the terms of the GNU Lesser General Public 11 * License as published by the Free Software Foundation; either 12 * version 2.1 of the License, or (at your option) any later version. 13 * 14 * FFmpeg is distributed in the hope that it will be useful, 15 * but WITHOUT ANY WARRANTY; without even the implied warranty of 16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 17 * Lesser General Public License for more details. 18 * 19 * You should have received a copy of the GNU Lesser General Public 20 * License along with FFmpeg; if not, write to the Free Software 21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 22 */ 23 24/** 25 * @file 26 * Direct Stream Digital (DSD) decoder 27 */ 28 29#include "libavcodec/internal.h" 30#include "libavcodec/mathops.h" 31#include "avcodec.h" 32#include "dsd_tablegen.h" 33 34#define FIFOSIZE 16 /** must be a power of two */ 35#define FIFOMASK (FIFOSIZE - 1) /** bit mask for FIFO offsets */ 36 37#if FIFOSIZE * 8 < HTAPS * 2 38#error "FIFOSIZE too small" 39#endif 40 41/** 42 * Per-channel buffer 43 */ 44typedef struct { 45 unsigned char buf[FIFOSIZE]; 46 unsigned pos; 47} DSDContext; 48 49static void dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf, 50 const unsigned char *src, ptrdiff_t src_stride, 51 float *dst, ptrdiff_t dst_stride) 52{ 53 unsigned pos, i; 54 unsigned char* p; 55 double sum; 56 57 pos = s->pos; 58 59 while (samples-- > 0) { 60 s->buf[pos] = lsbf ? ff_reverse[*src] : *src; 61 src += src_stride; 62 63 p = s->buf + ((pos - CTABLES) & FIFOMASK); 64 *p = ff_reverse[*p]; 65 66 sum = 0.0; 67 for (i = 0; i < CTABLES; i++) { 68 unsigned char a = s->buf[(pos - i) & FIFOMASK]; 69 unsigned char b = s->buf[(pos - (CTABLES*2 - 1) + i) & FIFOMASK]; 70 sum += ctables[i][a] + ctables[i][b]; 71 } 72 73 *dst = (float)sum; 74 dst += dst_stride; 75 76 pos = (pos + 1) & FIFOMASK; 77 } 78 79 s->pos = pos; 80} 81 82static av_cold void init_static_data(void) 83{ 84 static int done = 0; 85 if (done) 86 return; 87 dsd_ctables_tableinit(); 88 done = 1; 89} 90 91static av_cold int decode_init(AVCodecContext *avctx) 92{ 93 DSDContext * s; 94 int i; 95 96 init_static_data(); 97 98 s = av_malloc_array(sizeof(DSDContext), avctx->channels); 99 if (!s) 100 return AVERROR(ENOMEM); 101 102 for (i = 0; i < avctx->channels; i++) { 103 s[i].pos = 0; 104 memset(s[i].buf, 0x69, sizeof(s[i].buf)); 105 106 /* 0x69 = 01101001 107 * This pattern "on repeat" makes a low energy 352.8 kHz tone 108 * and a high energy 1.0584 MHz tone which should be filtered 109 * out completely by any playback system --> silence 110 */ 111 } 112 113 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; 114 avctx->priv_data = s; 115 return 0; 116} 117 118static int decode_frame(AVCodecContext *avctx, void *data, 119 int *got_frame_ptr, AVPacket *avpkt) 120{ 121 DSDContext * s = avctx->priv_data; 122 AVFrame *frame = data; 123 int ret, i; 124 int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR; 125 int src_next; 126 int src_stride; 127 128 frame->nb_samples = avpkt->size / avctx->channels; 129 130 if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) { 131 src_next = frame->nb_samples; 132 src_stride = 1; 133 } else { 134 src_next = 1; 135 src_stride = avctx->channels; 136 } 137 138 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 139 return ret; 140 141 for (i = 0; i < avctx->channels; i++) { 142 float * dst = ((float **)frame->extended_data)[i]; 143 dsd2pcm_translate(&s[i], frame->nb_samples, lsbf, 144 avpkt->data + i * src_next, src_stride, 145 dst, 1); 146 } 147 148 *got_frame_ptr = 1; 149 return frame->nb_samples * avctx->channels; 150} 151 152#define DSD_DECODER(id_, name_, long_name_) \ 153AVCodec ff_##name_##_decoder = { \ 154 .name = #name_, \ 155 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ 156 .type = AVMEDIA_TYPE_AUDIO, \ 157 .id = AV_CODEC_ID_##id_, \ 158 .init = decode_init, \ 159 .decode = decode_frame, \ 160 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \ 161 AV_SAMPLE_FMT_NONE }, \ 162}; 163 164DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first") 165DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first") 166DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar") 167DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar") 168