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