Commit | Line | Data |
---|---|---|
2ba45a60 DM |
1 | /* |
2 | * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> | |
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 <stdint.h> | |
22 | #include <string.h> | |
23 | ||
24 | #include "libavutil/mem.h" | |
25 | #include "audio_data.h" | |
26 | ||
27 | static const AVClass audio_data_class = { | |
28 | .class_name = "AudioData", | |
29 | .item_name = av_default_item_name, | |
30 | .version = LIBAVUTIL_VERSION_INT, | |
31 | }; | |
32 | ||
33 | /* | |
34 | * Calculate alignment for data pointers. | |
35 | */ | |
36 | static void calc_ptr_alignment(AudioData *a) | |
37 | { | |
38 | int p; | |
39 | int min_align = 128; | |
40 | ||
41 | for (p = 0; p < a->planes; p++) { | |
42 | int cur_align = 128; | |
43 | while ((intptr_t)a->data[p] % cur_align) | |
44 | cur_align >>= 1; | |
45 | if (cur_align < min_align) | |
46 | min_align = cur_align; | |
47 | } | |
48 | a->ptr_align = min_align; | |
49 | } | |
50 | ||
51 | int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels) | |
52 | { | |
53 | if (channels == 1) | |
54 | return 1; | |
55 | else | |
56 | return av_sample_fmt_is_planar(sample_fmt); | |
57 | } | |
58 | ||
59 | int ff_audio_data_set_channels(AudioData *a, int channels) | |
60 | { | |
61 | if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS || | |
62 | channels > a->allocated_channels) | |
63 | return AVERROR(EINVAL); | |
64 | ||
65 | a->channels = channels; | |
66 | a->planes = a->is_planar ? channels : 1; | |
67 | ||
68 | calc_ptr_alignment(a); | |
69 | ||
70 | return 0; | |
71 | } | |
72 | ||
73 | int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels, | |
74 | int nb_samples, enum AVSampleFormat sample_fmt, | |
75 | int read_only, const char *name) | |
76 | { | |
77 | int p; | |
78 | ||
79 | memset(a, 0, sizeof(*a)); | |
80 | a->class = &audio_data_class; | |
81 | ||
82 | if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) { | |
83 | av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels); | |
84 | return AVERROR(EINVAL); | |
85 | } | |
86 | ||
87 | a->sample_size = av_get_bytes_per_sample(sample_fmt); | |
88 | if (!a->sample_size) { | |
89 | av_log(a, AV_LOG_ERROR, "invalid sample format\n"); | |
90 | return AVERROR(EINVAL); | |
91 | } | |
92 | a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels); | |
93 | a->planes = a->is_planar ? channels : 1; | |
94 | a->stride = a->sample_size * (a->is_planar ? 1 : channels); | |
95 | ||
96 | for (p = 0; p < (a->is_planar ? channels : 1); p++) { | |
97 | if (!src[p]) { | |
98 | av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p); | |
99 | return AVERROR(EINVAL); | |
100 | } | |
101 | a->data[p] = src[p]; | |
102 | } | |
103 | a->allocated_samples = nb_samples * !read_only; | |
104 | a->nb_samples = nb_samples; | |
105 | a->sample_fmt = sample_fmt; | |
106 | a->channels = channels; | |
107 | a->allocated_channels = channels; | |
108 | a->read_only = read_only; | |
109 | a->allow_realloc = 0; | |
110 | a->name = name ? name : "{no name}"; | |
111 | ||
112 | calc_ptr_alignment(a); | |
113 | a->samples_align = plane_size / a->stride; | |
114 | ||
115 | return 0; | |
116 | } | |
117 | ||
118 | AudioData *ff_audio_data_alloc(int channels, int nb_samples, | |
119 | enum AVSampleFormat sample_fmt, const char *name) | |
120 | { | |
121 | AudioData *a; | |
122 | int ret; | |
123 | ||
124 | if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) | |
125 | return NULL; | |
126 | ||
127 | a = av_mallocz(sizeof(*a)); | |
128 | if (!a) | |
129 | return NULL; | |
130 | ||
131 | a->sample_size = av_get_bytes_per_sample(sample_fmt); | |
132 | if (!a->sample_size) { | |
133 | av_free(a); | |
134 | return NULL; | |
135 | } | |
136 | a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels); | |
137 | a->planes = a->is_planar ? channels : 1; | |
138 | a->stride = a->sample_size * (a->is_planar ? 1 : channels); | |
139 | ||
140 | a->class = &audio_data_class; | |
141 | a->sample_fmt = sample_fmt; | |
142 | a->channels = channels; | |
143 | a->allocated_channels = channels; | |
144 | a->read_only = 0; | |
145 | a->allow_realloc = 1; | |
146 | a->name = name ? name : "{no name}"; | |
147 | ||
148 | if (nb_samples > 0) { | |
149 | ret = ff_audio_data_realloc(a, nb_samples); | |
150 | if (ret < 0) { | |
151 | av_free(a); | |
152 | return NULL; | |
153 | } | |
154 | return a; | |
155 | } else { | |
156 | calc_ptr_alignment(a); | |
157 | return a; | |
158 | } | |
159 | } | |
160 | ||
161 | int ff_audio_data_realloc(AudioData *a, int nb_samples) | |
162 | { | |
163 | int ret, new_buf_size, plane_size, p; | |
164 | ||
165 | /* check if buffer is already large enough */ | |
166 | if (a->allocated_samples >= nb_samples) | |
167 | return 0; | |
168 | ||
169 | /* validate that the output is not read-only and realloc is allowed */ | |
170 | if (a->read_only || !a->allow_realloc) | |
171 | return AVERROR(EINVAL); | |
172 | ||
173 | new_buf_size = av_samples_get_buffer_size(&plane_size, | |
174 | a->allocated_channels, nb_samples, | |
175 | a->sample_fmt, 0); | |
176 | if (new_buf_size < 0) | |
177 | return new_buf_size; | |
178 | ||
179 | /* if there is already data in the buffer and the sample format is planar, | |
180 | allocate a new buffer and copy the data, otherwise just realloc the | |
181 | internal buffer and set new data pointers */ | |
182 | if (a->nb_samples > 0 && a->is_planar) { | |
183 | uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL }; | |
184 | ||
185 | ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels, | |
186 | nb_samples, a->sample_fmt, 0); | |
187 | if (ret < 0) | |
188 | return ret; | |
189 | ||
190 | for (p = 0; p < a->planes; p++) | |
191 | memcpy(new_data[p], a->data[p], a->nb_samples * a->stride); | |
192 | ||
193 | av_freep(&a->buffer); | |
194 | memcpy(a->data, new_data, sizeof(new_data)); | |
195 | a->buffer = a->data[0]; | |
196 | } else { | |
197 | av_freep(&a->buffer); | |
198 | a->buffer = av_malloc(new_buf_size); | |
199 | if (!a->buffer) | |
200 | return AVERROR(ENOMEM); | |
201 | ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer, | |
202 | a->allocated_channels, nb_samples, | |
203 | a->sample_fmt, 0); | |
204 | if (ret < 0) | |
205 | return ret; | |
206 | } | |
207 | a->buffer_size = new_buf_size; | |
208 | a->allocated_samples = nb_samples; | |
209 | ||
210 | calc_ptr_alignment(a); | |
211 | a->samples_align = plane_size / a->stride; | |
212 | ||
213 | return 0; | |
214 | } | |
215 | ||
216 | void ff_audio_data_free(AudioData **a) | |
217 | { | |
218 | if (!*a) | |
219 | return; | |
220 | av_free((*a)->buffer); | |
221 | av_freep(a); | |
222 | } | |
223 | ||
224 | int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map) | |
225 | { | |
226 | int ret, p; | |
227 | ||
228 | /* validate input/output compatibility */ | |
229 | if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels) | |
230 | return AVERROR(EINVAL); | |
231 | ||
232 | if (map && !src->is_planar) { | |
233 | av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n"); | |
234 | return AVERROR(EINVAL); | |
235 | } | |
236 | ||
237 | /* if the input is empty, just empty the output */ | |
238 | if (!src->nb_samples) { | |
239 | dst->nb_samples = 0; | |
240 | return 0; | |
241 | } | |
242 | ||
243 | /* reallocate output if necessary */ | |
244 | ret = ff_audio_data_realloc(dst, src->nb_samples); | |
245 | if (ret < 0) | |
246 | return ret; | |
247 | ||
248 | /* copy data */ | |
249 | if (map) { | |
250 | if (map->do_remap) { | |
251 | for (p = 0; p < src->planes; p++) { | |
252 | if (map->channel_map[p] >= 0) | |
253 | memcpy(dst->data[p], src->data[map->channel_map[p]], | |
254 | src->nb_samples * src->stride); | |
255 | } | |
256 | } | |
257 | if (map->do_copy || map->do_zero) { | |
258 | for (p = 0; p < src->planes; p++) { | |
259 | if (map->channel_copy[p]) | |
260 | memcpy(dst->data[p], dst->data[map->channel_copy[p]], | |
261 | src->nb_samples * src->stride); | |
262 | else if (map->channel_zero[p]) | |
263 | av_samples_set_silence(&dst->data[p], 0, src->nb_samples, | |
264 | 1, dst->sample_fmt); | |
265 | } | |
266 | } | |
267 | } else { | |
268 | for (p = 0; p < src->planes; p++) | |
269 | memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride); | |
270 | } | |
271 | ||
272 | dst->nb_samples = src->nb_samples; | |
273 | ||
274 | return 0; | |
275 | } | |
276 | ||
277 | int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, | |
278 | int src_offset, int nb_samples) | |
279 | { | |
280 | int ret, p, dst_offset2, dst_move_size; | |
281 | ||
282 | /* validate input/output compatibility */ | |
283 | if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) { | |
284 | av_log(src, AV_LOG_ERROR, "sample format mismatch\n"); | |
285 | return AVERROR(EINVAL); | |
286 | } | |
287 | ||
288 | /* validate offsets are within the buffer bounds */ | |
289 | if (dst_offset < 0 || dst_offset > dst->nb_samples || | |
290 | src_offset < 0 || src_offset > src->nb_samples) { | |
291 | av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n", | |
292 | src_offset, dst_offset); | |
293 | return AVERROR(EINVAL); | |
294 | } | |
295 | ||
296 | /* check offsets and sizes to see if we can just do nothing and return */ | |
297 | if (nb_samples > src->nb_samples - src_offset) | |
298 | nb_samples = src->nb_samples - src_offset; | |
299 | if (nb_samples <= 0) | |
300 | return 0; | |
301 | ||
302 | /* validate that the output is not read-only */ | |
303 | if (dst->read_only) { | |
304 | av_log(dst, AV_LOG_ERROR, "dst is read-only\n"); | |
305 | return AVERROR(EINVAL); | |
306 | } | |
307 | ||
308 | /* reallocate output if necessary */ | |
309 | ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples); | |
310 | if (ret < 0) { | |
311 | av_log(dst, AV_LOG_ERROR, "error reallocating dst\n"); | |
312 | return ret; | |
313 | } | |
314 | ||
315 | dst_offset2 = dst_offset + nb_samples; | |
316 | dst_move_size = dst->nb_samples - dst_offset; | |
317 | ||
318 | for (p = 0; p < src->planes; p++) { | |
319 | if (dst_move_size > 0) { | |
320 | memmove(dst->data[p] + dst_offset2 * dst->stride, | |
321 | dst->data[p] + dst_offset * dst->stride, | |
322 | dst_move_size * dst->stride); | |
323 | } | |
324 | memcpy(dst->data[p] + dst_offset * dst->stride, | |
325 | src->data[p] + src_offset * src->stride, | |
326 | nb_samples * src->stride); | |
327 | } | |
328 | dst->nb_samples += nb_samples; | |
329 | ||
330 | return 0; | |
331 | } | |
332 | ||
333 | void ff_audio_data_drain(AudioData *a, int nb_samples) | |
334 | { | |
335 | if (a->nb_samples <= nb_samples) { | |
336 | /* drain the whole buffer */ | |
337 | a->nb_samples = 0; | |
338 | } else { | |
339 | int p; | |
340 | int move_offset = a->stride * nb_samples; | |
341 | int move_size = a->stride * (a->nb_samples - nb_samples); | |
342 | ||
343 | for (p = 0; p < a->planes; p++) | |
344 | memmove(a->data[p], a->data[p] + move_offset, move_size); | |
345 | ||
346 | a->nb_samples -= nb_samples; | |
347 | } | |
348 | } | |
349 | ||
350 | int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, | |
351 | int nb_samples) | |
352 | { | |
353 | uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS]; | |
354 | int offset_size, p; | |
355 | ||
356 | if (offset >= a->nb_samples) | |
357 | return 0; | |
358 | offset_size = offset * a->stride; | |
359 | for (p = 0; p < a->planes; p++) | |
360 | offset_data[p] = a->data[p] + offset_size; | |
361 | ||
362 | return av_audio_fifo_write(af, (void **)offset_data, nb_samples); | |
363 | } | |
364 | ||
365 | int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples) | |
366 | { | |
367 | int ret; | |
368 | ||
369 | if (a->read_only) | |
370 | return AVERROR(EINVAL); | |
371 | ||
372 | ret = ff_audio_data_realloc(a, nb_samples); | |
373 | if (ret < 0) | |
374 | return ret; | |
375 | ||
376 | ret = av_audio_fifo_read(af, (void **)a->data, nb_samples); | |
377 | if (ret >= 0) | |
378 | a->nb_samples = ret; | |
379 | return ret; | |
380 | } |