Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / dsddec.c
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")