FFmpeg coverage

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