1 /*
2 * Copyright (c) 2012 Laurent Aimar
3 *
4 * This file is part of FFmpeg.
5 *
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/intreadwrite.h"
22 #include "avcodec.h"
23 #include "internal.h"
24 #include "dvaudio.h"
25
26 typedef struct DVAudioContext {
27 int block_size;
28 int is_12bit;
29 int is_pal;
30 int16_t shuffle[2000];
31 } DVAudioContext;
32
decode_init(AVCodecContext * avctx)33 static av_cold int decode_init(AVCodecContext *avctx)
34 {
35 DVAudioContext *s = avctx->priv_data;
36 int i;
37
38 if (avctx->channels != 2) {
39 av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
40 return AVERROR(EINVAL);
41 }
42
43 if (avctx->codec_tag == 0x0215) {
44 s->block_size = 7200;
45 } else if (avctx->codec_tag == 0x0216) {
46 s->block_size = 8640;
47 } else if (avctx->block_align == 7200 ||
48 avctx->block_align == 8640) {
49 s->block_size = avctx->block_align;
50 } else {
51 return AVERROR(EINVAL);
52 }
53
54 s->is_pal = s->block_size == 8640;
55 s->is_12bit = avctx->bits_per_coded_sample == 12;
56 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
57 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
58
59 for (i = 0; i < FF_ARRAY_ELEMS(s->shuffle); i++) {
60 const unsigned a = s->is_pal ? 18 : 15;
61 const unsigned b = 3 * a;
62
63 s->shuffle[i] = 80 * ((21 * (i % 3) + 9 * (i / 3) + ((i / a) % 3)) % b) +
64 (2 + s->is_12bit) * (i / b) + 8;
65 }
66
67 return 0;
68 }
69
dv_audio_12to16(uint16_t sample)70 static inline uint16_t dv_audio_12to16(uint16_t sample)
71 {
72 uint16_t shift, result;
73
74 sample = (sample < 0x800) ? sample : sample | 0xf000;
75 shift = (sample & 0xf00) >> 8;
76
77 if (shift < 0x2 || shift > 0xd) {
78 result = sample;
79 } else if (shift < 0x8) {
80 shift--;
81 result = (sample - (256 * shift)) << shift;
82 } else {
83 shift = 0xe - shift;
84 result = ((sample + ((256 * shift) + 1)) << shift) - 1;
85 }
86
87 return result;
88 }
89
decode_frame(AVCodecContext * avctx,void * data,int * got_frame_ptr,AVPacket * pkt)90 static int decode_frame(AVCodecContext *avctx, void *data,
91 int *got_frame_ptr, AVPacket *pkt)
92 {
93 DVAudioContext *s = avctx->priv_data;
94 AVFrame *frame = data;
95 const uint8_t *src = pkt->data;
96 int16_t *dst;
97 int ret, i;
98
99 if (pkt->size < s->block_size)
100 return AVERROR_INVALIDDATA;
101
102 frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244, s->is_pal);
103 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
104 return ret;
105 dst = (int16_t *)frame->data[0];
106
107 for (i = 0; i < frame->nb_samples; i++) {
108 const uint8_t *v = &src[s->shuffle[i]];
109
110 if (s->is_12bit) {
111 *dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f));
112 *dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f));
113 } else {
114 *dst++ = AV_RB16(&v[0]);
115 *dst++ = AV_RB16(&v[s->is_pal ? 4320 : 3600]);
116 }
117 }
118
119 *got_frame_ptr = 1;
120
121 return s->block_size;
122 }
123
124 AVCodec ff_dvaudio_decoder = {
125 .name = "dvaudio",
126 .long_name = NULL_IF_CONFIG_SMALL("Ulead DV Audio"),
127 .type = AVMEDIA_TYPE_AUDIO,
128 .id = AV_CODEC_ID_DVAUDIO,
129 .init = decode_init,
130 .decode = decode_frame,
131 .capabilities = AV_CODEC_CAP_DR1,
132 .priv_data_size = sizeof(DVAudioContext),
133 };
134