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/channel_layout.h"
22 #include "libavutil/intreadwrite.h"
23 #include "avcodec.h"
24 #include "codec_internal.h"
25 #include "internal.h"
26 #include "dvaudio.h"
27
28 typedef struct DVAudioContext {
29 int block_size;
30 int is_12bit;
31 int is_pal;
32 int16_t shuffle[2000];
33 } DVAudioContext;
34
decode_init(AVCodecContext * avctx)35 static av_cold int decode_init(AVCodecContext *avctx)
36 {
37 DVAudioContext *s = avctx->priv_data;
38 int i;
39
40 if (avctx->codec_tag == 0x0215) {
41 s->block_size = 7200;
42 } else if (avctx->codec_tag == 0x0216) {
43 s->block_size = 8640;
44 } else if (avctx->block_align == 7200 ||
45 avctx->block_align == 8640) {
46 s->block_size = avctx->block_align;
47 } else {
48 return AVERROR(EINVAL);
49 }
50
51 s->is_pal = s->block_size == 8640;
52 s->is_12bit = avctx->bits_per_coded_sample == 12;
53 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
54 av_channel_layout_uninit(&avctx->ch_layout);
55 avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
56
57 for (i = 0; i < FF_ARRAY_ELEMS(s->shuffle); i++) {
58 const unsigned a = s->is_pal ? 18 : 15;
59 const unsigned b = 3 * a;
60
61 s->shuffle[i] = 80 * ((21 * (i % 3) + 9 * (i / 3) + ((i / a) % 3)) % b) +
62 (2 + s->is_12bit) * (i / b) + 8;
63 }
64
65 return 0;
66 }
67
dv_audio_12to16(uint16_t sample)68 static inline uint16_t dv_audio_12to16(uint16_t sample)
69 {
70 uint16_t shift, result;
71
72 sample = (sample < 0x800) ? sample : sample | 0xf000;
73 shift = (sample & 0xf00) >> 8;
74
75 if (shift < 0x2 || shift > 0xd) {
76 result = sample;
77 } else if (shift < 0x8) {
78 shift--;
79 result = (sample - (256 * shift)) << shift;
80 } else {
81 shift = 0xe - shift;
82 result = ((sample + ((256 * shift) + 1)) << shift) - 1;
83 }
84
85 return result;
86 }
87
decode_frame(AVCodecContext * avctx,AVFrame * frame,int * got_frame_ptr,AVPacket * pkt)88 static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
89 int *got_frame_ptr, AVPacket *pkt)
90 {
91 DVAudioContext *s = avctx->priv_data;
92 const uint8_t *src = pkt->data;
93 int16_t *dst;
94 int ret, i;
95
96 if (pkt->size < s->block_size)
97 return AVERROR_INVALIDDATA;
98
99 frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244, s->is_pal);
100 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
101 return ret;
102 dst = (int16_t *)frame->data[0];
103
104 for (i = 0; i < frame->nb_samples; i++) {
105 const uint8_t *v = &src[s->shuffle[i]];
106
107 if (s->is_12bit) {
108 *dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f));
109 *dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f));
110 } else {
111 *dst++ = AV_RB16(&v[0]);
112 *dst++ = AV_RB16(&v[s->is_pal ? 4320 : 3600]);
113 }
114 }
115
116 *got_frame_ptr = 1;
117
118 return s->block_size;
119 }
120
121 const FFCodec ff_dvaudio_decoder = {
122 .p.name = "dvaudio",
123 .p.long_name = NULL_IF_CONFIG_SMALL("Ulead DV Audio"),
124 .p.type = AVMEDIA_TYPE_AUDIO,
125 .p.id = AV_CODEC_ID_DVAUDIO,
126 .init = decode_init,
127 FF_CODEC_DECODE_CB(decode_frame),
128 .p.capabilities = AV_CODEC_CAP_DR1,
129 .priv_data_size = sizeof(DVAudioContext),
130 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
131 };
132