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/internal.h"
23			#include "libavutil/intreadwrite.h"
24			#include "avcodec.h"
25			#include "codec_internal.h"
26			#include "decode.h"
27			#include "get_bits.h"
28
29			typedef struct ChContext {
30			int have_code;
31			int last_sample;
32			int last_delta;
33			int bit_length;
34			int block_length;
35			uint8_t block[32 * 2];
36			AVAudioFifo *samples;
37			} ChContext;
38
39			typedef struct APACContext {
40			GetBitContext gb;
41			int skip;
42
43			int cur_ch;
44			ChContext ch[2];
45
46			uint8_t *bitstream;
47			int64_t max_framesize;
48			int bitstream_size;
49			int bitstream_index;
50			} APACContext;
51
52		✗	static av_cold int apac_close(AVCodecContext *avctx)
53			{
54		✗	APACContext *s = avctx->priv_data;
55
56		✗	av_freep(&s->bitstream);
57		✗	s->bitstream_size = 0;
58
59		✗	for (int ch = 0; ch < 2; ch++) {
60		✗	ChContext *c = &s->ch[ch];
61
62		✗	av_audio_fifo_free(c->samples);
63			}
64
65		✗	return 0;
66			}
67
68		✗	static av_cold int apac_init(AVCodecContext *avctx)
69			{
70		✗	APACContext *s = avctx->priv_data;
71
72		✗	if (avctx->bits_per_coded_sample > 8)
73		✗	avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
74			else
75		✗	avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
76
77		✗	if (avctx->ch_layout.nb_channels < 1 ||
78		✗	avctx->ch_layout.nb_channels > 2 ||
79		✗	avctx->bits_per_coded_sample < 8 ||
80		✗	avctx->bits_per_coded_sample > 16
81			)
82		✗	return AVERROR_INVALIDDATA;
83
84		✗	for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
85		✗	ChContext *c = &s->ch[ch];
86
87		✗	c->bit_length = avctx->bits_per_coded_sample;
88		✗	c->block_length = 8;
89		✗	c->have_code = 0;
90		✗	c->samples = av_audio_fifo_alloc(avctx->sample_fmt, 1, 1024);
91		✗	if (!c->samples)
92		✗	return AVERROR(ENOMEM);
93			}
94
95		✗	s->max_framesize = 1024;
96		✗	s->bitstream = av_realloc_f(s->bitstream, s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));
97		✗	if (!s->bitstream)
98		✗	return AVERROR(ENOMEM);
99
100		✗	return 0;
101			}
102
103		✗	static int get_code(ChContext *c, GetBitContext *gb)
104			{
105		✗	if (get_bits1(gb)) {
106		✗	int code = get_bits(gb, 2);
107
108		✗	switch (code) {
109		✗	case 0:
110		✗	c->bit_length--;
111		✗	break;
112		✗	case 1:
113		✗	c->bit_length++;
114		✗	break;
115		✗	case 2:
116		✗	c->bit_length = get_bits(gb, 5);
117		✗	break;
118		✗	case 3:
119		✗	c->block_length = get_bits(gb, 4);
120		✗	return 1;
121			}
122			}
123
124		✗	return 0;
125			}
126
127		✗	static int apac_decode(AVCodecContext *avctx, AVFrame *frame,
128			int *got_frame_ptr, AVPacket *pkt)
129			{
130		✗	APACContext *s = avctx->priv_data;
131		✗	GetBitContext *gb = &s->gb;
132			int ret, n, buf_size, input_buf_size;
133			const uint8_t *buf;
134			int nb_samples;
135
136		✗	if (!pkt->size && s->bitstream_size <= 0) {
137		✗	*got_frame_ptr = 0;
138		✗	return 0;
139			}
140
141		✗	buf_size = pkt->size;
142		✗	input_buf_size = buf_size;
143
144		✗	if (s->bitstream_index > 0 && s->bitstream_size > 0) {
145		✗	memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
146		✗	s->bitstream_index = 0;
147			}
148
149		✗	if (s->bitstream_index + s->bitstream_size + buf_size > s->max_framesize) {
150		✗	s->bitstream = av_realloc_f(s->bitstream, s->bitstream_index +
151		✗	s->bitstream_size +
152		✗	buf_size + AV_INPUT_BUFFER_PADDING_SIZE,
153			sizeof(*s->bitstream));
154		✗	if (!s->bitstream)
155		✗	return AVERROR(ENOMEM);
156		✗	s->max_framesize = s->bitstream_index + s->bitstream_size + buf_size;
157			}
158		✗	if (pkt->data)
159		✗	memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);
160		✗	buf = &s->bitstream[s->bitstream_index];
161		✗	buf_size += s->bitstream_size;
162		✗	s->bitstream_size = buf_size;
163
164		✗	frame->nb_samples = s->bitstream_size * 16 * 8;
165		✗	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
166		✗	return ret;
167
168		✗	if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)
169		✗	return ret;
170
171		✗	skip_bits(gb, s->skip);
172		✗	s->skip = 0;
173
174		✗	while (get_bits_left(gb) > 0) {
175		✗	for (int ch = s->cur_ch; ch < avctx->ch_layout.nb_channels; ch++) {
176		✗	ChContext *c = &s->ch[ch];
177		✗	int16_t *dst16 = (int16_t *)c->block;
178		✗	uint8_t *dst8 = (uint8_t *)c->block;
179			void *samples[4];
180
181		✗	samples[0] = &c->block[0];
182		✗	if (get_bits_left(gb) < 16 && pkt->size) {
183		✗	s->cur_ch = ch;
184		✗	goto end;
185			}
186
187		✗	if (!c->have_code && get_code(c, gb))
188		✗	get_code(c, gb);
189		✗	c->have_code = 0;
190
191		✗	if (c->block_length <= 0)
192		✗	continue;
193
194		✗	if (c->bit_length < 0 ||
195		✗	c->bit_length > 17) {
196		✗	c->bit_length = avctx->bits_per_coded_sample;
197		✗	s->bitstream_index = 0;
198		✗	s->bitstream_size = 0;
199		✗	return AVERROR_INVALIDDATA;
200			}
201
202		✗	if (get_bits_left(gb) < c->block_length * c->bit_length) {
203		✗	if (pkt->size) {
204		✗	c->have_code = 1;
205		✗	s->cur_ch = ch;
206		✗	goto end;
207			} else {
208		✗	break;
209			}
210			}
211
212		✗	for (int i = 0; i < c->block_length; i++) {
213		✗	int val = get_bits_long(gb, c->bit_length);
214		✗	unsigned delta = (val & 1) ? ~(val >> 1) : (val >> 1);
215			int sample;
216
217		✗	delta += c->last_delta;
218		✗	sample = c->last_sample + delta;
219		✗	c->last_delta = delta;
220		✗	c->last_sample = sample;
221
222		✗	switch (avctx->sample_fmt) {
223		✗	case AV_SAMPLE_FMT_S16P:
224		✗	dst16[i] = sample;
225		✗	break;
226		✗	case AV_SAMPLE_FMT_U8P:
227		✗	dst8[i] = sample;
228		✗	break;
229			}
230			}
231
232		✗	av_audio_fifo_write(c->samples, samples, c->block_length);
233			}
234
235		✗	s->cur_ch = 0;
236			}
237		✗	end:
238		✗	nb_samples = frame->nb_samples;
239		✗	for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
240		✗	nb_samples = FFMIN(av_audio_fifo_size(s->ch[ch].samples), nb_samples);
241
242		✗	frame->nb_samples = nb_samples;
243		✗	for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
244		✗	void *samples[1] = { frame->extended_data[ch] };
245		✗	av_audio_fifo_read(s->ch[ch].samples, samples, nb_samples);
246			}
247
248		✗	s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
249		✗	n = get_bits_count(gb) / 8;
250
251		✗	if (nb_samples > 0 || pkt->size)
252		✗	*got_frame_ptr = 1;
253
254		✗	if (s->bitstream_size > 0) {
255		✗	s->bitstream_index += n;
256		✗	s->bitstream_size -= n;
257		✗	return input_buf_size;
258			}
259		✗	return n;
260			}
261
262			const FFCodec ff_apac_decoder = {
263			.p.name = "apac",
264			CODEC_LONG_NAME("Marian's A-pac audio"),
265			.p.type = AVMEDIA_TYPE_AUDIO,
266			.p.id = AV_CODEC_ID_APAC,
267			.priv_data_size = sizeof(APACContext),
268			.init = apac_init,
269			FF_CODEC_DECODE_CB(apac_decode),
270			.close = apac_close,
271			.p.capabilities = AV_CODEC_CAP_DELAY |
272			#if FF_API_SUBFRAMES
273			AV_CODEC_CAP_SUBFRAMES |
274			#endif
275			AV_CODEC_CAP_DR1,
276			.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
277			.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
278			AV_SAMPLE_FMT_S16P,
279			AV_SAMPLE_FMT_NONE },
280			};
281