| 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 | } |