| 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 | */ |
| 44 | typedef struct { |
| 45 | unsigned char buf[FIFOSIZE]; |
| 46 | unsigned pos; |
| 47 | } DSDContext; |
| 48 | |
| 49 | static 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 | |
| 82 | static 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 | |
| 91 | static 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 | |
| 118 | static 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_) \ |
| 153 | AVCodec 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 | |
| 164 | DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first") |
| 165 | DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first") |
| 166 | DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar") |
| 167 | DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar") |