FFmpeg
vima.c
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1 /*
2  * LucasArts VIMA decoder
3  * Copyright (c) 2012 Paul B Mahol
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * LucasArts VIMA audio decoder
25  * @author Paul B Mahol
26  */
27 
28 #include "libavutil/channel_layout.h"
29 
30 #include "adpcm_data.h"
31 #include "avcodec.h"
32 #include "get_bits.h"
33 #include "internal.h"
34 
35 static int predict_table_init = 0;
36 static uint16_t predict_table[5786 * 2];
37 
38 static const uint8_t size_table[] = {
39  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
40  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
41  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
42  5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
43  6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
44  7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7
45 };
46 
47 static const int8_t index_table1[] = {
48  -1, 4, -1, 4
49 };
50 
51 static const int8_t index_table2[] = {
52  -1, -1, 2, 6, -1, -1, 2, 6
53 };
54 
55 static const int8_t index_table3[] = {
56  -1, -1, -1, -1, 1, 2, 4, 6, -1, -1, -1, -1, 1, 2, 4, 6
57 };
58 
59 static const int8_t index_table4[] = {
60  -1, -1, -1, -1, -1, -1, -1, -1, 1, 1, 1, 2, 2, 4, 5, 6,
61  -1, -1, -1, -1, -1, -1, -1, -1, 1, 1, 1, 2, 2, 4, 5, 6
62 };
63 
64 static const int8_t index_table5[] = {
65  -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
66  1, 1, 1, 1, 1, 2, 2, 2, 2, 4, 4, 4, 5, 5, 6, 6,
67  -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
68  1, 1, 1, 1, 1, 2, 2, 2, 2, 4, 4, 4, 5, 5, 6, 6
69 };
70 
71 static const int8_t index_table6[] = {
72  -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
73  -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
74  1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2,
75  2, 2, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 6, 6, 6, 6,
76  -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
77  -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
78  1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2,
79  2, 2, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 6, 6, 6, 6
80 };
81 
82 static const int8_t *const step_index_tables[] = {
83  index_table1, index_table2, index_table3,
84  index_table4, index_table5, index_table6
85 };
86 
87 static av_cold int decode_init(AVCodecContext *avctx)
88 {
89  int start_pos;
90 
91  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
92 
93  if (predict_table_init)
94  return 0;
95 
96  for (start_pos = 0; start_pos < 64; start_pos++) {
97  unsigned int dest_pos, table_pos;
98 
99  for (table_pos = 0, dest_pos = start_pos;
100  table_pos < FF_ARRAY_ELEMS(ff_adpcm_step_table);
101  table_pos++, dest_pos += 64) {
102  int put = 0, count, table_value;
103 
104  table_value = ff_adpcm_step_table[table_pos];
105  for (count = 32; count != 0; count >>= 1) {
106  if (start_pos & count)
107  put += table_value;
108  table_value >>= 1;
109  }
110  predict_table[dest_pos] = put;
111  }
112  }
113  predict_table_init = 1;
114 
115  return 0;
116 }
117 
118 static int decode_frame(AVCodecContext *avctx, void *data,
119  int *got_frame_ptr, AVPacket *pkt)
120 {
121  GetBitContext gb;
122  AVFrame *frame = data;
123  int16_t pcm_data[2];
124  uint32_t samples;
125  int8_t channel_hint[2];
126  int ret, chan;
127  int channels = 1;
128 
129  if (pkt->size < 13)
130  return AVERROR_INVALIDDATA;
131 
132  if ((ret = init_get_bits8(&gb, pkt->data, pkt->size)) < 0)
133  return ret;
134 
135  samples = get_bits_long(&gb, 32);
136  if (samples == 0xffffffff) {
137  skip_bits_long(&gb, 32);
138  samples = get_bits_long(&gb, 32);
139  }
140 
141  if (samples > pkt->size * 2)
142  return AVERROR_INVALIDDATA;
143 
144  channel_hint[0] = get_sbits(&gb, 8);
145  if (channel_hint[0] & 0x80) {
146  channel_hint[0] = ~channel_hint[0];
147  channels = 2;
148  }
149  avctx->channels = channels;
150  avctx->channel_layout = (channels == 2) ? AV_CH_LAYOUT_STEREO
151  : AV_CH_LAYOUT_MONO;
152  pcm_data[0] = get_sbits(&gb, 16);
153  if (channels > 1) {
154  channel_hint[1] = get_sbits(&gb, 8);
155  pcm_data[1] = get_sbits(&gb, 16);
156  }
157 
158  frame->nb_samples = samples;
159  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
160  return ret;
161 
162  for (chan = 0; chan < channels; chan++) {
163  uint16_t *dest = (uint16_t *)frame->data[0] + chan;
164  int step_index = channel_hint[chan];
165  int output = pcm_data[chan];
166  int sample;
167 
168  for (sample = 0; sample < samples; sample++) {
169  int lookup_size, lookup, highbit, lowbits;
170 
171  step_index = av_clip(step_index, 0, 88);
172  lookup_size = size_table[step_index];
173  lookup = get_bits(&gb, lookup_size);
174  highbit = 1 << (lookup_size - 1);
175  lowbits = highbit - 1;
176 
177  if (lookup & highbit)
178  lookup ^= highbit;
179  else
180  highbit = 0;
181 
182  if (lookup == lowbits) {
183  output = get_sbits(&gb, 16);
184  } else {
185  int predict_index, diff;
186 
187  predict_index = (lookup << (7 - lookup_size)) | (step_index << 6);
188  predict_index = av_clip(predict_index, 0, 5785);
189  diff = predict_table[predict_index];
190  if (lookup)
191  diff += ff_adpcm_step_table[step_index] >> (lookup_size - 1);
192  if (highbit)
193  diff = -diff;
194 
195  output = av_clip_int16(output + diff);
196  }
197 
198  *dest = output;
199  dest += channels;
200 
201  step_index += step_index_tables[lookup_size - 2][lookup];
202  }
203  }
204 
205  *got_frame_ptr = 1;
206 
207  return pkt->size;
208 }
209 
210 AVCodec ff_adpcm_vima_decoder = {
211  .name = "adpcm_vima",
212  .long_name = NULL_IF_CONFIG_SMALL("LucasArts VIMA audio"),
213  .type = AVMEDIA_TYPE_AUDIO,
214  .id = AV_CODEC_ID_ADPCM_VIMA,
215  .init = decode_init,
216  .decode = decode_frame,
217  .capabilities = AV_CODEC_CAP_DR1,
218 };
AVCodec
AVCodec.
Definition: codec.h:190
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
get_bits_long
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:546
output
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce output
Definition: filter_design.txt:225
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:355
data
const char data[16]
Definition: mxf.c:91
ff_adpcm_vima_decoder
AVCodec ff_adpcm_vima_decoder
Definition: vima.c:210
index_table2
static const int8_t index_table2[]
Definition: vima.c:51
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
index_table4
static const int8_t index_table4[]
Definition: vima.c:59
GetBitContext
Definition: get_bits.h:61
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:86
av_cold
#define av_cold
Definition: attributes.h:90
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
adpcm_data.h
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
get_sbits
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:359
channels
channels
Definition: aptx.h:33
get_bits.h
index_table5
static const int8_t index_table5[]
Definition: vima.c:64
index_table1
static const int8_t index_table1[]
Definition: vima.c:47
predict_table_init
static int predict_table_init
Definition: vima.c:35
AV_CODEC_ID_ADPCM_VIMA
@ AV_CODEC_ID_ADPCM_VIMA
Definition: codec_id.h:370
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
decode_frame
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *pkt)
Definition: vima.c:118
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1854
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
AVPacket::size
int size
Definition: packet.h:356
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:188
ff_adpcm_step_table
const int16_t ff_adpcm_step_table[89]
This is the step table.
Definition: adpcm_data.c:61
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
sample
#define sample
Definition: flacdsp_template.c:44
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
index_table6
static const int8_t index_table6[]
Definition: vima.c:71
lookup
int lookup
Definition: vorbis_enc_data.h:455
step_index_tables
static const int8_t *const step_index_tables[]
Definition: vima.c:82
uint8_t
uint8_t
Definition: audio_convert.c:194
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:197
avcodec.h
ret
ret
Definition: filter_design.txt:187
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen_template.c:38
predict_table
static uint16_t predict_table[5786 *2]
Definition: vima.c:36
AVCodecContext
main external API structure.
Definition: avcodec.h:526
channel_layout.h
pkt
static AVPacket pkt
Definition: demuxing_decoding.c:54
index_table3
static const int8_t index_table3[]
Definition: vima.c:55
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
diff
static av_always_inline int diff(const uint32_t a, const uint32_t b)
Definition: vf_palettegen.c:136
AVPacket
This structure stores compressed data.
Definition: packet.h:332
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
size_table
static const uint8_t size_table[]
Definition: vima.c:38
decode_init
static av_cold int decode_init(AVCodecContext *avctx)
Definition: vima.c:87
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