FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/apac.c
Date: 2024-04-20 14:10:07
Exec Total Coverage
Lines: 0 137 0.0%
Functions: 0 4 0.0%
Branches: 0 84 0.0%

Line Branch Exec Source
1 /*
2 * APAC audio decoder
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/audio_fifo.h"
22 #include "libavutil/mem.h"
23 #include "avcodec.h"
24 #include "codec_internal.h"
25 #include "decode.h"
26 #include "get_bits.h"
27
28 typedef struct ChContext {
29 int have_code;
30 int last_sample;
31 int last_delta;
32 int bit_length;
33 int block_length;
34 uint8_t block[32 * 2];
35 AVAudioFifo *samples;
36 } ChContext;
37
38 typedef struct APACContext {
39 GetBitContext gb;
40 int skip;
41
42 int cur_ch;
43 ChContext ch[2];
44
45 uint8_t *bitstream;
46 int64_t max_framesize;
47 int bitstream_size;
48 int bitstream_index;
49 } APACContext;
50
51 static av_cold int apac_close(AVCodecContext *avctx)
52 {
53 APACContext *s = avctx->priv_data;
54
55 av_freep(&s->bitstream);
56 s->bitstream_size = 0;
57
58 for (int ch = 0; ch < 2; ch++) {
59 ChContext *c = &s->ch[ch];
60
61 av_audio_fifo_free(c->samples);
62 }
63
64 return 0;
65 }
66
67 static av_cold int apac_init(AVCodecContext *avctx)
68 {
69 APACContext *s = avctx->priv_data;
70
71 if (avctx->bits_per_coded_sample > 8)
72 avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
73 else
74 avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
75
76 if (avctx->ch_layout.nb_channels < 1 ||
77 avctx->ch_layout.nb_channels > 2 ||
78 avctx->bits_per_coded_sample < 8 ||
79 avctx->bits_per_coded_sample > 16
80 )
81 return AVERROR_INVALIDDATA;
82
83 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
84 ChContext *c = &s->ch[ch];
85
86 c->bit_length = avctx->bits_per_coded_sample;
87 c->block_length = 8;
88 c->have_code = 0;
89 c->samples = av_audio_fifo_alloc(avctx->sample_fmt, 1, 1024);
90 if (!c->samples)
91 return AVERROR(ENOMEM);
92 }
93
94 s->max_framesize = 1024;
95 s->bitstream = av_realloc_f(s->bitstream, s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));
96 if (!s->bitstream)
97 return AVERROR(ENOMEM);
98
99 return 0;
100 }
101
102 static int get_code(ChContext *c, GetBitContext *gb)
103 {
104 if (get_bits1(gb)) {
105 int code = get_bits(gb, 2);
106
107 switch (code) {
108 case 0:
109 c->bit_length--;
110 break;
111 case 1:
112 c->bit_length++;
113 break;
114 case 2:
115 c->bit_length = get_bits(gb, 5);
116 break;
117 case 3:
118 c->block_length = get_bits(gb, 4);
119 return 1;
120 }
121 }
122
123 return 0;
124 }
125
126 static int apac_decode(AVCodecContext *avctx, AVFrame *frame,
127 int *got_frame_ptr, AVPacket *pkt)
128 {
129 APACContext *s = avctx->priv_data;
130 GetBitContext *gb = &s->gb;
131 int ret, n, buf_size, input_buf_size;
132 const uint8_t *buf;
133 int nb_samples;
134
135 if (!pkt->size && s->bitstream_size <= 0) {
136 *got_frame_ptr = 0;
137 return 0;
138 }
139
140 buf_size = pkt->size;
141 input_buf_size = buf_size;
142
143 if (s->bitstream_index > 0 && s->bitstream_size > 0) {
144 memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
145 s->bitstream_index = 0;
146 }
147
148 if (s->bitstream_index + s->bitstream_size + buf_size > s->max_framesize) {
149 s->bitstream = av_realloc_f(s->bitstream, s->bitstream_index +
150 s->bitstream_size +
151 buf_size + AV_INPUT_BUFFER_PADDING_SIZE,
152 sizeof(*s->bitstream));
153 if (!s->bitstream)
154 return AVERROR(ENOMEM);
155 s->max_framesize = s->bitstream_index + s->bitstream_size + buf_size;
156 }
157 if (pkt->data)
158 memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);
159 buf = &s->bitstream[s->bitstream_index];
160 buf_size += s->bitstream_size;
161 s->bitstream_size = buf_size;
162
163 frame->nb_samples = s->bitstream_size * 16 * 8;
164 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
165 return ret;
166
167 if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)
168 return ret;
169
170 skip_bits(gb, s->skip);
171 s->skip = 0;
172
173 while (get_bits_left(gb) > 0) {
174 for (int ch = s->cur_ch; ch < avctx->ch_layout.nb_channels; ch++) {
175 ChContext *c = &s->ch[ch];
176 int16_t *dst16 = (int16_t *)c->block;
177 uint8_t *dst8 = (uint8_t *)c->block;
178 void *samples[4];
179
180 samples[0] = &c->block[0];
181 if (get_bits_left(gb) < 16 && pkt->size) {
182 s->cur_ch = ch;
183 goto end;
184 }
185
186 if (!c->have_code && get_code(c, gb))
187 get_code(c, gb);
188 c->have_code = 0;
189
190 if (c->block_length <= 0)
191 continue;
192
193 if (c->bit_length < 0 ||
194 c->bit_length > 17) {
195 c->bit_length = avctx->bits_per_coded_sample;
196 s->bitstream_index = 0;
197 s->bitstream_size = 0;
198 return AVERROR_INVALIDDATA;
199 }
200
201 if (get_bits_left(gb) < c->block_length * c->bit_length) {
202 if (pkt->size) {
203 c->have_code = 1;
204 s->cur_ch = ch;
205 goto end;
206 } else {
207 break;
208 }
209 }
210
211 for (int i = 0; i < c->block_length; i++) {
212 int val = get_bits_long(gb, c->bit_length);
213 unsigned delta = (val & 1) ? ~(val >> 1) : (val >> 1);
214 int sample;
215
216 delta += c->last_delta;
217 sample = c->last_sample + delta;
218 c->last_delta = delta;
219 c->last_sample = sample;
220
221 switch (avctx->sample_fmt) {
222 case AV_SAMPLE_FMT_S16P:
223 dst16[i] = sample;
224 break;
225 case AV_SAMPLE_FMT_U8P:
226 dst8[i] = sample;
227 break;
228 }
229 }
230
231 av_audio_fifo_write(c->samples, samples, c->block_length);
232 }
233
234 s->cur_ch = 0;
235 }
236 end:
237 nb_samples = frame->nb_samples;
238 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
239 nb_samples = FFMIN(av_audio_fifo_size(s->ch[ch].samples), nb_samples);
240
241 frame->nb_samples = nb_samples;
242 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
243 void *samples[1] = { frame->extended_data[ch] };
244 av_audio_fifo_read(s->ch[ch].samples, samples, nb_samples);
245 }
246
247 s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
248 n = get_bits_count(gb) / 8;
249
250 if (nb_samples > 0 || pkt->size)
251 *got_frame_ptr = 1;
252
253 if (s->bitstream_size > 0) {
254 s->bitstream_index += n;
255 s->bitstream_size -= n;
256 return input_buf_size;
257 }
258 return n;
259 }
260
261 const FFCodec ff_apac_decoder = {
262 .p.name = "apac",
263 CODEC_LONG_NAME("Marian's A-pac audio"),
264 .p.type = AVMEDIA_TYPE_AUDIO,
265 .p.id = AV_CODEC_ID_APAC,
266 .priv_data_size = sizeof(APACContext),
267 .init = apac_init,
268 FF_CODEC_DECODE_CB(apac_decode),
269 .close = apac_close,
270 .p.capabilities = AV_CODEC_CAP_DELAY |
271 #if FF_API_SUBFRAMES
272 AV_CODEC_CAP_SUBFRAMES |
273 #endif
274 AV_CODEC_CAP_DR1,
275 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
276 .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
277 AV_SAMPLE_FMT_S16P,
278 AV_SAMPLE_FMT_NONE },
279 };
280