| @@ -0,0 +1,811 @@ |
| | 1 | /* |
| | 2 | * Copyright (C) 2017 Paul B Mahol |
| | 3 | * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda |
| | 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 <math.h> |
| | 22 | |
| | 23 | #include "libavutil/audio_fifo.h" |
| | 24 | #include "libavutil/avstring.h" |
| | 25 | #include "libavutil/channel_layout.h" |
| | 26 | #include "libavutil/float_dsp.h" |
| | 27 | #include "libavutil/intmath.h" |
| | 28 | #include "libavutil/opt.h" |
| | 29 | #include "libavcodec/avfft.h" |
| | 30 | |
| | 31 | #include "avfilter.h" |
| | 32 | #include "internal.h" |
| | 33 | #include "audio.h" |
| | 34 | |
| | 35 | #define TIME_DOMAIN 0 |
| | 36 | #define FREQUENCY_DOMAIN 1 |
| | 37 | |
| | 38 | typedef struct HeadphoneContext { |
| | 39 | const AVClass *class; |
| | 40 | |
| | 41 | char *map; |
| | 42 | int type; |
| | 43 | |
| | 44 | int lfe_channel; |
| | 45 | |
| | 46 | int have_hrirs; |
| | 47 | int eof_hrirs; |
| | 48 | int64_t pts; |
| | 49 | |
| | 50 | int ir_len; |
| | 51 | |
| | 52 | int mapping[64]; |
| | 53 | |
| | 54 | int nb_inputs; |
| | 55 | |
| | 56 | int nb_irs; |
| | 57 | |
| | 58 | float gain; |
| | 59 | float lfe_gain, gain_lfe; |
| | 60 | |
| | 61 | float *ringbuffer[2]; |
| | 62 | int write[2]; |
| | 63 | |
| | 64 | int buffer_length; |
| | 65 | int n_fft; |
| | 66 | int size; |
| | 67 | |
| | 68 | int *delay[2]; |
| | 69 | float *data_ir[2]; |
| | 70 | float *temp_src[2]; |
| | 71 | FFTComplex *temp_fft[2]; |
| | 72 | |
| | 73 | FFTContext *fft[2], *ifft[2]; |
| | 74 | FFTComplex *data_hrtf[2]; |
| | 75 | |
| | 76 | AVFloatDSPContext *fdsp; |
| | 77 | struct headphone_inputs { |
| | 78 | AVAudioFifo *fifo; |
| | 79 | AVFrame *frame; |
| | 80 | int ir_len; |
| | 81 | int delay_l; |
| | 82 | int delay_r; |
| | 83 | int eof; |
| | 84 | } *in; |
| | 85 | } HeadphoneContext; |
| | 86 | |
| | 87 | static int parse_channel_name(HeadphoneContext *s, int x, char **arg, int *rchannel, char *buf) |
| | 88 | { |
| | 89 | int len, i, channel_id = 0; |
| | 90 | int64_t layout, layout0; |
| | 91 | |
| | 92 | if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) { |
| | 93 | layout0 = layout = av_get_channel_layout(buf); |
| | 94 | if (layout == AV_CH_LOW_FREQUENCY) |
| | 95 | s->lfe_channel = x; |
| | 96 | for (i = 32; i > 0; i >>= 1) { |
| | 97 | if (layout >= 1LL << i) { |
| | 98 | channel_id += i; |
| | 99 | layout >>= i; |
| | 100 | } |
| | 101 | } |
| | 102 | if (channel_id >= 64 || layout0 != 1LL << channel_id) |
| | 103 | return AVERROR(EINVAL); |
| | 104 | *rchannel = channel_id; |
| | 105 | *arg += len; |
| | 106 | return 0; |
| | 107 | } |
| | 108 | return AVERROR(EINVAL); |
| | 109 | } |
| | 110 | |
| | 111 | static void parse_map(AVFilterContext *ctx) |
| | 112 | { |
| | 113 | HeadphoneContext *s = ctx->priv; |
| | 114 | char *arg, *tokenizer, *p, *args = av_strdup(s->map); |
| | 115 | int i; |
| | 116 | |
| | 117 | if (!args) |
| | 118 | return; |
| | 119 | p = args; |
| | 120 | |
| | 121 | s->lfe_channel = -1; |
| | 122 | s->nb_inputs = 1; |
| | 123 | |
| | 124 | for (i = 0; i < 64; i++) { |
| | 125 | s->mapping[i] = -1; |
| | 126 | } |
| | 127 | |
| | 128 | while ((arg = av_strtok(p, "|", &tokenizer))) { |
| | 129 | int out_ch_id; |
| | 130 | char buf[8]; |
| | 131 | |
| | 132 | p = NULL; |
| | 133 | if (parse_channel_name(s, s->nb_inputs - 1, &arg, &out_ch_id, buf)) { |
| | 134 | av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf); |
| | 135 | continue; |
| | 136 | } |
| | 137 | s->mapping[s->nb_inputs - 1] = out_ch_id; |
| | 138 | s->nb_inputs++; |
| | 139 | } |
| | 140 | s->nb_irs = s->nb_inputs - 1; |
| | 141 | |
| | 142 | av_free(args); |
| | 143 | } |
| | 144 | |
| | 145 | typedef struct ThreadData { |
| | 146 | AVFrame *in, *out; |
| | 147 | int *write; |
| | 148 | int **delay; |
| | 149 | float **ir; |
| | 150 | int *n_clippings; |
| | 151 | float **ringbuffer; |
| | 152 | float **temp_src; |
| | 153 | FFTComplex **temp_fft; |
| | 154 | } ThreadData; |
| | 155 | |
| | 156 | static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
| | 157 | { |
| | 158 | HeadphoneContext *s = ctx->priv; |
| | 159 | ThreadData *td = arg; |
| | 160 | AVFrame *in = td->in, *out = td->out; |
| | 161 | int offset = jobnr; |
| | 162 | int *write = &td->write[jobnr]; |
| | 163 | const int *const delay = td->delay[jobnr]; |
| | 164 | const float *const ir = td->ir[jobnr]; |
| | 165 | int *n_clippings = &td->n_clippings[jobnr]; |
| | 166 | float *ringbuffer = td->ringbuffer[jobnr]; |
| | 167 | float *temp_src = td->temp_src[jobnr]; |
| | 168 | const int ir_len = s->ir_len; |
| | 169 | const float *src = (const float *)in->data[0]; |
| | 170 | float *dst = (float *)out->data[0]; |
| | 171 | const int in_channels = in->channels; |
| | 172 | const int buffer_length = s->buffer_length; |
| | 173 | const uint32_t modulo = (uint32_t)buffer_length - 1; |
| | 174 | float *buffer[16]; |
| | 175 | int wr = *write; |
| | 176 | int read; |
| | 177 | int i, l; |
| | 178 | |
| | 179 | dst += offset; |
| | 180 | for (l = 0; l < in_channels; l++) { |
| | 181 | buffer[l] = ringbuffer + l * buffer_length; |
| | 182 | } |
| | 183 | |
| | 184 | for (i = 0; i < in->nb_samples; i++) { |
| | 185 | const float *temp_ir = ir; |
| | 186 | |
| | 187 | *dst = 0; |
| | 188 | for (l = 0; l < in_channels; l++) { |
| | 189 | *(buffer[l] + wr) = src[l]; |
| | 190 | } |
| | 191 | |
| | 192 | for (l = 0; l < in_channels; l++) { |
| | 193 | const float *const bptr = buffer[l]; |
| | 194 | |
| | 195 | if (l == s->lfe_channel) { |
| | 196 | *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe; |
| | 197 | temp_ir += FFALIGN(ir_len, 16); |
| | 198 | continue; |
| | 199 | } |
| | 200 | |
| | 201 | read = (wr - *(delay + l) - (ir_len - 1) + buffer_length) & modulo; |
| | 202 | |
| | 203 | if (read + ir_len < buffer_length) { |
| | 204 | memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src)); |
| | 205 | } else { |
| | 206 | int len = FFMIN(ir_len - (read % ir_len), buffer_length - read); |
| | 207 | |
| | 208 | memcpy(temp_src, bptr + read, len * sizeof(*temp_src)); |
| | 209 | memcpy(temp_src + len, bptr, (ir_len - len) * sizeof(*temp_src)); |
| | 210 | } |
| | 211 | |
| | 212 | dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, ir_len); |
| | 213 | temp_ir += FFALIGN(ir_len, 16); |
| | 214 | } |
| | 215 | |
| | 216 | if (fabs(*dst) > 1) |
| | 217 | *n_clippings += 1; |
| | 218 | |
| | 219 | dst += 2; |
| | 220 | src += in_channels; |
| | 221 | wr = (wr + 1) & modulo; |
| | 222 | } |
| | 223 | |
| | 224 | *write = wr; |
| | 225 | |
| | 226 | return 0; |
| | 227 | } |
| | 228 | |
| | 229 | static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
| | 230 | { |
| | 231 | HeadphoneContext *s = ctx->priv; |
| | 232 | ThreadData *td = arg; |
| | 233 | AVFrame *in = td->in, *out = td->out; |
| | 234 | int offset = jobnr; |
| | 235 | int *write = &td->write[jobnr]; |
| | 236 | FFTComplex *hrtf = s->data_hrtf[jobnr]; |
| | 237 | int *n_clippings = &td->n_clippings[jobnr]; |
| | 238 | float *ringbuffer = td->ringbuffer[jobnr]; |
| | 239 | const int ir_len = s->ir_len; |
| | 240 | const float *src = (const float *)in->data[0]; |
| | 241 | float *dst = (float *)out->data[0]; |
| | 242 | const int in_channels = in->channels; |
| | 243 | const int buffer_length = s->buffer_length; |
| | 244 | const uint32_t modulo = (uint32_t)buffer_length - 1; |
| | 245 | FFTComplex *fft_in = s->temp_fft[jobnr]; |
| | 246 | FFTContext *ifft = s->ifft[jobnr]; |
| | 247 | FFTContext *fft = s->fft[jobnr]; |
| | 248 | const int n_fft = s->n_fft; |
| | 249 | const float fft_scale = 1.0f / s->n_fft; |
| | 250 | FFTComplex *hrtf_offset; |
| | 251 | int wr = *write; |
| | 252 | int n_read; |
| | 253 | int i, j; |
| | 254 | |
| | 255 | dst += offset; |
| | 256 | |
| | 257 | n_read = FFMIN(s->ir_len, in->nb_samples); |
| | 258 | for (j = 0; j < n_read; j++) { |
| | 259 | dst[2 * j] = ringbuffer[wr]; |
| | 260 | ringbuffer[wr] = 0.0; |
| | 261 | wr = (wr + 1) & modulo; |
| | 262 | } |
| | 263 | |
| | 264 | for (j = n_read; j < in->nb_samples; j++) { |
| | 265 | dst[2 * j] = 0; |
| | 266 | } |
| | 267 | |
| | 268 | for (i = 0; i < in_channels; i++) { |
| | 269 | if (i == s->lfe_channel) { |
| | 270 | for (j = 0; j < in->nb_samples; j++) { |
| | 271 | dst[2 * j] += src[i + j * in_channels] * s->gain_lfe; |
| | 272 | } |
| | 273 | continue; |
| | 274 | } |
| | 275 | |
| | 276 | offset = i * n_fft; |
| | 277 | hrtf_offset = hrtf + offset; |
| | 278 | |
| | 279 | memset(fft_in, 0, sizeof(FFTComplex) * n_fft); |
| | 280 | |
| | 281 | for (j = 0; j < in->nb_samples; j++) { |
| | 282 | fft_in[j].re = src[j * in_channels + i]; |
| | 283 | } |
| | 284 | |
| | 285 | av_fft_permute(fft, fft_in); |
| | 286 | av_fft_calc(fft, fft_in); |
| | 287 | for (j = 0; j < n_fft; j++) { |
| | 288 | const FFTComplex *hcomplex = hrtf_offset + j; |
| | 289 | const float re = fft_in[j].re; |
| | 290 | const float im = fft_in[j].im; |
| | 291 | |
| | 292 | fft_in[j].re = re * hcomplex->re - im * hcomplex->im; |
| | 293 | fft_in[j].im = re * hcomplex->im + im * hcomplex->re; |
| | 294 | } |
| | 295 | |
| | 296 | av_fft_permute(ifft, fft_in); |
| | 297 | av_fft_calc(ifft, fft_in); |
| | 298 | |
| | 299 | for (j = 0; j < in->nb_samples; j++) { |
| | 300 | dst[2 * j] += fft_in[j].re * fft_scale; |
| | 301 | } |
| | 302 | |
| | 303 | for (j = 0; j < ir_len - 1; j++) { |
| | 304 | int write_pos = (wr + j) & modulo; |
| | 305 | |
| | 306 | *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale; |
| | 307 | } |
| | 308 | } |
| | 309 | |
| | 310 | for (i = 0; i < out->nb_samples; i++) { |
| | 311 | if (fabs(*dst) > 1) { |
| | 312 | n_clippings[0]++; |
| | 313 | } |
| | 314 | |
| | 315 | dst += 2; |
| | 316 | } |
| | 317 | |
| | 318 | *write = wr; |
| | 319 | |
| | 320 | return 0; |
| | 321 | } |
| | 322 | |
| | 323 | static int read_ir(AVFilterLink *inlink, AVFrame *frame) |
| | 324 | { |
| | 325 | AVFilterContext *ctx = inlink->dst; |
| | 326 | HeadphoneContext *s = ctx->priv; |
| | 327 | int ir_len, max_ir_len, input_number; |
| | 328 | |
| | 329 | for (input_number = 0; input_number < s->nb_inputs; input_number++) |
| | 330 | if (inlink == ctx->inputs[input_number]) |
| | 331 | break; |
| | 332 | |
| | 333 | av_audio_fifo_write(s->in[input_number].fifo, (void **)frame->extended_data, |
| | 334 | frame->nb_samples); |
| | 335 | av_frame_free(&frame); |
| | 336 | |
| | 337 | ir_len = av_audio_fifo_size(s->in[input_number].fifo); |
| | 338 | max_ir_len = 4096; |
| | 339 | if (ir_len > max_ir_len) { |
| | 340 | av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len); |
| | 341 | return AVERROR(EINVAL); |
| | 342 | } |
| | 343 | s->in[input_number].ir_len = ir_len; |
| | 344 | s->ir_len = FFMAX(ir_len, s->ir_len); |
| | 345 | |
| | 346 | return 0; |
| | 347 | } |
| | 348 | |
| | 349 | static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink) |
| | 350 | { |
| | 351 | AVFilterContext *ctx = outlink->src; |
| | 352 | AVFrame *in = s->in[0].frame; |
| | 353 | int n_clippings[2] = { 0 }; |
| | 354 | ThreadData td; |
| | 355 | AVFrame *out; |
| | 356 | |
| | 357 | av_audio_fifo_read(s->in[0].fifo, (void **)in->extended_data, s->size); |
| | 358 | |
| | 359 | out = ff_get_audio_buffer(outlink, in->nb_samples); |
| | 360 | if (!out) { |
| | 361 | av_frame_free(&in); |
| | 362 | return AVERROR(ENOMEM); |
| | 363 | } |
| | 364 | out->pts = s->pts; |
| | 365 | if (s->pts != AV_NOPTS_VALUE) |
| | 366 | s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
| | 367 | |
| | 368 | td.in = in; td.out = out; td.write = s->write; |
| | 369 | td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings; |
| | 370 | td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src; |
| | 371 | td.temp_fft = s->temp_fft; |
| | 372 | |
| | 373 | if (s->type == TIME_DOMAIN) { |
| | 374 | ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2); |
| | 375 | } else { |
| | 376 | ctx->internal->execute(ctx, headphone_fast_convolute, &td, NULL, 2); |
| | 377 | } |
| | 378 | emms_c(); |
| | 379 | |
| | 380 | if (n_clippings[0] + n_clippings[1] > 0) { |
| | 381 | av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n", |
| | 382 | n_clippings[0] + n_clippings[1], out->nb_samples * 2); |
| | 383 | } |
| | 384 | |
| | 385 | return ff_filter_frame(outlink, out); |
| | 386 | } |
| | 387 | |
| | 388 | static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink) |
| | 389 | { |
| | 390 | struct HeadphoneContext *s = ctx->priv; |
| | 391 | const int ir_len = s->ir_len; |
| | 392 | int nb_irs = s->nb_irs; |
| | 393 | int nb_input_channels = ctx->inputs[0]->channels; |
| | 394 | float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); |
| | 395 | FFTComplex *data_hrtf_l = NULL; |
| | 396 | FFTComplex *data_hrtf_r = NULL; |
| | 397 | FFTComplex *fft_in_l = NULL; |
| | 398 | FFTComplex *fft_in_r = NULL; |
| | 399 | float *data_ir_l = NULL; |
| | 400 | float *data_ir_r = NULL; |
| | 401 | int offset = 0; |
| | 402 | int n_fft; |
| | 403 | int i, j; |
| | 404 | |
| | 405 | s->buffer_length = 1 << (32 - ff_clz(s->ir_len)); |
| | 406 | s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + inlink->sample_rate)); |
| | 407 | |
| | 408 | if (s->type == FREQUENCY_DOMAIN) { |
| | 409 | fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l)); |
| | 410 | fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r)); |
| | 411 | if (!fft_in_l || !fft_in_r) { |
| | 412 | return AVERROR(ENOMEM); |
| | 413 | } |
| | 414 | |
| | 415 | av_fft_end(s->fft[0]); |
| | 416 | av_fft_end(s->fft[1]); |
| | 417 | s->fft[0] = av_fft_init(log2(s->n_fft), 0); |
| | 418 | s->fft[1] = av_fft_init(log2(s->n_fft), 0); |
| | 419 | av_fft_end(s->ifft[0]); |
| | 420 | av_fft_end(s->ifft[1]); |
| | 421 | s->ifft[0] = av_fft_init(log2(s->n_fft), 1); |
| | 422 | s->ifft[1] = av_fft_init(log2(s->n_fft), 1); |
| | 423 | |
| | 424 | if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) { |
| | 425 | av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft); |
| | 426 | return AVERROR(ENOMEM); |
| | 427 | } |
| | 428 | } |
| | 429 | |
| | 430 | s->data_ir[0] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs); |
| | 431 | s->data_ir[1] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs); |
| | 432 | s->delay[0] = av_malloc_array(s->nb_irs, sizeof(float)); |
| | 433 | s->delay[1] = av_malloc_array(s->nb_irs, sizeof(float)); |
| | 434 | |
| | 435 | if (s->type == TIME_DOMAIN) { |
| | 436 | s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); |
| | 437 | s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); |
| | 438 | } else { |
| | 439 | s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float)); |
| | 440 | s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float)); |
| | 441 | s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); |
| | 442 | s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); |
| | 443 | if (!s->temp_fft[0] || !s->temp_fft[1]) |
| | 444 | return AVERROR(ENOMEM); |
| | 445 | } |
| | 446 | |
| | 447 | if (!s->data_ir[0] || !s->data_ir[1] || |
| | 448 | !s->ringbuffer[0] || !s->ringbuffer[1]) |
| | 449 | return AVERROR(ENOMEM); |
| | 450 | |
| | 451 | s->in[0].frame = ff_get_audio_buffer(ctx->inputs[0], s->size); |
| | 452 | if (!s->in[0].frame) |
| | 453 | return AVERROR(ENOMEM); |
| | 454 | for (i = 0; i < s->nb_irs; i++) { |
| | 455 | s->in[i + 1].frame = ff_get_audio_buffer(ctx->inputs[i + 1], s->ir_len); |
| | 456 | if (!s->in[i + 1].frame) |
| | 457 | return AVERROR(ENOMEM); |
| | 458 | } |
| | 459 | |
| | 460 | if (s->type == TIME_DOMAIN) { |
| | 461 | s->temp_src[0] = av_calloc(FFALIGN(ir_len, 16), sizeof(float)); |
| | 462 | s->temp_src[1] = av_calloc(FFALIGN(ir_len, 16), sizeof(float)); |
| | 463 | |
| | 464 | data_ir_l = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_l)); |
| | 465 | data_ir_r = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_r)); |
| | 466 | if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) { |
| | 467 | av_free(data_ir_l); |
| | 468 | av_free(data_ir_r); |
| | 469 | return AVERROR(ENOMEM); |
| | 470 | } |
| | 471 | } else { |
| | 472 | data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * nb_irs); |
| | 473 | data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * nb_irs); |
| | 474 | if (!data_hrtf_r || !data_hrtf_l) { |
| | 475 | av_free(data_hrtf_l); |
| | 476 | av_free(data_hrtf_r); |
| | 477 | return AVERROR(ENOMEM); |
| | 478 | } |
| | 479 | } |
| | 480 | |
| | 481 | for (i = 0; i < s->nb_irs; i++) { |
| | 482 | int len = s->in[i + 1].ir_len; |
| | 483 | int delay_l = s->in[i + 1].delay_l; |
| | 484 | int delay_r = s->in[i + 1].delay_r; |
| | 485 | int idx = -1; |
| | 486 | float *ptr; |
| | 487 | |
| | 488 | for (j = 0; j < inlink->channels; j++) { |
| | 489 | if (s->mapping[i] < 0) { |
| | 490 | continue; |
| | 491 | } |
| | 492 | |
| | 493 | if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[i])) { |
| | 494 | idx = j; |
| | 495 | break; |
| | 496 | } |
| | 497 | } |
| | 498 | if (idx == -1) |
| | 499 | continue; |
| | 500 | |
| | 501 | av_audio_fifo_read(s->in[i + 1].fifo, (void **)s->in[i + 1].frame->extended_data, len); |
| | 502 | ptr = (float *)s->in[i + 1].frame->extended_data[0]; |
| | 503 | |
| | 504 | if (s->type == TIME_DOMAIN) { |
| | 505 | offset = idx * FFALIGN(len, 16); |
| | 506 | for (j = 0; j < len; j++) { |
| | 507 | data_ir_l[offset + j] = ptr[len * 2 - j * 2 - 2] * gain_lin; |
| | 508 | data_ir_r[offset + j] = ptr[len * 2 - j * 2 - 1] * gain_lin; |
| | 509 | } |
| | 510 | } else { |
| | 511 | memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l)); |
| | 512 | memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r)); |
| | 513 | |
| | 514 | offset = idx * n_fft; |
| | 515 | for (j = 0; j < len; j++) { |
| | 516 | fft_in_l[delay_l + j].re = ptr[j * 2 ] * gain_lin; |
| | 517 | fft_in_r[delay_r + j].re = ptr[j * 2 + 1] * gain_lin; |
| | 518 | } |
| | 519 | |
| | 520 | av_fft_permute(s->fft[0], fft_in_l); |
| | 521 | av_fft_calc(s->fft[0], fft_in_l); |
| | 522 | memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l)); |
| | 523 | av_fft_permute(s->fft[0], fft_in_r); |
| | 524 | av_fft_calc(s->fft[0], fft_in_r); |
| | 525 | memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r)); |
| | 526 | } |
| | 527 | } |
| | 528 | |
| | 529 | if (s->type == TIME_DOMAIN) { |
| | 530 | memcpy(s->data_ir[0], data_ir_l, sizeof(float) * nb_irs * FFALIGN(ir_len, 16)); |
| | 531 | memcpy(s->data_ir[1], data_ir_r, sizeof(float) * nb_irs * FFALIGN(ir_len, 16)); |
| | 532 | |
| | 533 | av_freep(&data_ir_l); |
| | 534 | av_freep(&data_ir_r); |
| | 535 | } else { |
| | 536 | s->data_hrtf[0] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex)); |
| | 537 | s->data_hrtf[1] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex)); |
| | 538 | if (!s->data_hrtf[0] || !s->data_hrtf[1]) { |
| | 539 | av_freep(&data_hrtf_l); |
| | 540 | av_freep(&data_hrtf_r); |
| | 541 | av_freep(&fft_in_l); |
| | 542 | av_freep(&fft_in_r); |
| | 543 | return AVERROR(ENOMEM); |
| | 544 | } |
| | 545 | |
| | 546 | memcpy(s->data_hrtf[0], data_hrtf_l, |
| | 547 | sizeof(FFTComplex) * nb_irs * n_fft); |
| | 548 | memcpy(s->data_hrtf[1], data_hrtf_r, |
| | 549 | sizeof(FFTComplex) * nb_irs * n_fft); |
| | 550 | |
| | 551 | av_freep(&data_hrtf_l); |
| | 552 | av_freep(&data_hrtf_r); |
| | 553 | |
| | 554 | av_freep(&fft_in_l); |
| | 555 | av_freep(&fft_in_r); |
| | 556 | } |
| | 557 | |
| | 558 | s->have_hrirs = 1; |
| | 559 | |
| | 560 | return 0; |
| | 561 | } |
| | 562 | |
| | 563 | static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
| | 564 | { |
| | 565 | AVFilterContext *ctx = inlink->dst; |
| | 566 | HeadphoneContext *s = ctx->priv; |
| | 567 | AVFilterLink *outlink = ctx->outputs[0]; |
| | 568 | int ret = 0; |
| | 569 | |
| | 570 | av_audio_fifo_write(s->in[0].fifo, (void **)in->extended_data, |
| | 571 | in->nb_samples); |
| | 572 | if (s->pts == AV_NOPTS_VALUE) |
| | 573 | s->pts = in->pts; |
| | 574 | |
| | 575 | av_frame_free(&in); |
| | 576 | |
| | 577 | if (!s->have_hrirs && s->eof_hrirs) { |
| | 578 | ret = convert_coeffs(ctx, inlink); |
| | 579 | if (ret < 0) |
| | 580 | return ret; |
| | 581 | } |
| | 582 | |
| | 583 | if (s->have_hrirs) { |
| | 584 | while (av_audio_fifo_size(s->in[0].fifo) >= s->size) { |
| | 585 | ret = headphone_frame(s, outlink); |
| | 586 | if (ret < 0) |
| | 587 | break; |
| | 588 | } |
| | 589 | } |
| | 590 | return ret; |
| | 591 | } |
| | 592 | |
| | 593 | static int query_formats(AVFilterContext *ctx) |
| | 594 | { |
| | 595 | struct HeadphoneContext *s = ctx->priv; |
| | 596 | AVFilterFormats *formats = NULL; |
| | 597 | AVFilterChannelLayouts *layouts = NULL; |
| | 598 | int ret, i; |
| | 599 | |
| | 600 | ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT); |
| | 601 | if (ret) |
| | 602 | return ret; |
| | 603 | ret = ff_set_common_formats(ctx, formats); |
| | 604 | if (ret) |
| | 605 | return ret; |
| | 606 | |
| | 607 | layouts = ff_all_channel_layouts(); |
| | 608 | if (!layouts) |
| | 609 | return AVERROR(ENOMEM); |
| | 610 | |
| | 611 | ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts); |
| | 612 | if (ret) |
| | 613 | return ret; |
| | 614 | |
| | 615 | layouts = NULL; |
| | 616 | ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO); |
| | 617 | if (ret) |
| | 618 | return ret; |
| | 619 | |
| | 620 | for (i = 1; i < s->nb_inputs; i++) { |
| | 621 | ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts); |
| | 622 | if (ret) |
| | 623 | return ret; |
| | 624 | } |
| | 625 | |
| | 626 | ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts); |
| | 627 | if (ret) |
| | 628 | return ret; |
| | 629 | |
| | 630 | formats = ff_all_samplerates(); |
| | 631 | if (!formats) |
| | 632 | return AVERROR(ENOMEM); |
| | 633 | return ff_set_common_samplerates(ctx, formats); |
| | 634 | } |
| | 635 | |
| | 636 | static int config_input(AVFilterLink *inlink) |
| | 637 | { |
| | 638 | AVFilterContext *ctx = inlink->dst; |
| | 639 | HeadphoneContext *s = ctx->priv; |
| | 640 | |
| | 641 | if (s->type == FREQUENCY_DOMAIN) { |
| | 642 | inlink->partial_buf_size = |
| | 643 | inlink->min_samples = |
| | 644 | inlink->max_samples = inlink->sample_rate; |
| | 645 | } |
| | 646 | |
| | 647 | if (s->nb_irs < inlink->channels) { |
| | 648 | av_log(ctx, AV_LOG_ERROR, "Number of inputs must be >= %d.\n", inlink->channels + 1); |
| | 649 | return AVERROR(EINVAL); |
| | 650 | } |
| | 651 | |
| | 652 | return 0; |
| | 653 | } |
| | 654 | |
| | 655 | static av_cold int init(AVFilterContext *ctx) |
| | 656 | { |
| | 657 | HeadphoneContext *s = ctx->priv; |
| | 658 | int i; |
| | 659 | |
| | 660 | AVFilterPad pad = { |
| | 661 | .name = "in0", |
| | 662 | .type = AVMEDIA_TYPE_AUDIO, |
| | 663 | .config_props = config_input, |
| | 664 | .filter_frame = filter_frame, |
| | 665 | }; |
| | 666 | ff_insert_inpad(ctx, 0, &pad); |
| | 667 | |
| | 668 | if (!s->map) { |
| | 669 | av_log(ctx, AV_LOG_ERROR, "Valid mapping must be set.\n"); |
| | 670 | return AVERROR(EINVAL); |
| | 671 | } |
| | 672 | |
| | 673 | parse_map(ctx); |
| | 674 | |
| | 675 | s->in = av_calloc(s->nb_inputs, sizeof(*s->in)); |
| | 676 | if (!s->in) |
| | 677 | return AVERROR(ENOMEM); |
| | 678 | |
| | 679 | for (i = 1; i < s->nb_inputs; i++) { |
| | 680 | char *name = av_asprintf("hrir%d", i - 1); |
| | 681 | AVFilterPad pad = { |
| | 682 | .name = name, |
| | 683 | .type = AVMEDIA_TYPE_AUDIO, |
| | 684 | .filter_frame = read_ir, |
| | 685 | }; |
| | 686 | if (!name) |
| | 687 | return AVERROR(ENOMEM); |
| | 688 | ff_insert_inpad(ctx, i, &pad); |
| | 689 | } |
| | 690 | |
| | 691 | s->fdsp = avpriv_float_dsp_alloc(0); |
| | 692 | if (!s->fdsp) |
| | 693 | return AVERROR(ENOMEM); |
| | 694 | s->pts = AV_NOPTS_VALUE; |
| | 695 | |
| | 696 | return 0; |
| | 697 | } |
| | 698 | |
| | 699 | static int config_output(AVFilterLink *outlink) |
| | 700 | { |
| | 701 | AVFilterContext *ctx = outlink->src; |
| | 702 | HeadphoneContext *s = ctx->priv; |
| | 703 | AVFilterLink *inlink = ctx->inputs[0]; |
| | 704 | int i; |
| | 705 | |
| | 706 | if (s->type == TIME_DOMAIN) |
| | 707 | s->size = 1024; |
| | 708 | else |
| | 709 | s->size = inlink->sample_rate; |
| | 710 | |
| | 711 | for (i = 0; i < s->nb_inputs; i++) { |
| | 712 | s->in[i].fifo = av_audio_fifo_alloc(ctx->inputs[i]->format, ctx->inputs[i]->channels, 1024); |
| | 713 | if (!s->in[i].fifo) |
| | 714 | return AVERROR(ENOMEM); |
| | 715 | } |
| | 716 | s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6 + s->lfe_gain) / 20 * M_LN10); |
| | 717 | |
| | 718 | return 0; |
| | 719 | } |
| | 720 | |
| | 721 | static int request_frame(AVFilterLink *outlink) |
| | 722 | { |
| | 723 | AVFilterContext *ctx = outlink->src; |
| | 724 | HeadphoneContext *s = ctx->priv; |
| | 725 | int i, ret; |
| | 726 | |
| | 727 | for (i = 1; !s->eof_hrirs && i < s->nb_inputs; i++) { |
| | 728 | if (!s->in[i].eof) { |
| | 729 | ret = ff_request_frame(ctx->inputs[i]); |
| | 730 | if (ret == AVERROR_EOF) { |
| | 731 | s->in[i].eof = 1; |
| | 732 | ret = 0; |
| | 733 | } |
| | 734 | return ret; |
| | 735 | } else { |
| | 736 | if (i == s->nb_inputs - 1) |
| | 737 | s->eof_hrirs = 1; |
| | 738 | } |
| | 739 | } |
| | 740 | return ff_request_frame(ctx->inputs[0]); |
| | 741 | } |
| | 742 | |
| | 743 | static av_cold void uninit(AVFilterContext *ctx) |
| | 744 | { |
| | 745 | HeadphoneContext *s = ctx->priv; |
| | 746 | int i; |
| | 747 | |
| | 748 | av_fft_end(s->ifft[0]); |
| | 749 | av_fft_end(s->ifft[1]); |
| | 750 | av_fft_end(s->fft[0]); |
| | 751 | av_fft_end(s->fft[1]); |
| | 752 | av_freep(&s->delay[0]); |
| | 753 | av_freep(&s->delay[1]); |
| | 754 | av_freep(&s->data_ir[0]); |
| | 755 | av_freep(&s->data_ir[1]); |
| | 756 | av_freep(&s->ringbuffer[0]); |
| | 757 | av_freep(&s->ringbuffer[1]); |
| | 758 | av_freep(&s->temp_src[0]); |
| | 759 | av_freep(&s->temp_src[1]); |
| | 760 | av_freep(&s->temp_fft[0]); |
| | 761 | av_freep(&s->temp_fft[1]); |
| | 762 | av_freep(&s->data_hrtf[0]); |
| | 763 | av_freep(&s->data_hrtf[1]); |
| | 764 | av_freep(&s->fdsp); |
| | 765 | |
| | 766 | for (i = 0; i < s->nb_inputs; i++) { |
| | 767 | av_frame_free(&s->in[i].frame); |
| | 768 | av_audio_fifo_free(s->in[i].fifo); |
| | 769 | if (ctx->input_pads && i) |
| | 770 | av_freep(&ctx->input_pads[i].name); |
| | 771 | } |
| | 772 | av_freep(&s->in); |
| | 773 | } |
| | 774 | |
| | 775 | #define OFFSET(x) offsetof(HeadphoneContext, x) |
| | 776 | #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| | 777 | |
| | 778 | static const AVOption headphone_options[] = { |
| | 779 | { "map", "set channels convolution mappings", OFFSET(map), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS }, |
| | 780 | { "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS }, |
| | 781 | { "lfe", "set lfe gain in dB", OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS }, |
| | 782 | { "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" }, |
| | 783 | { "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" }, |
| | 784 | { "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" }, |
| | 785 | { NULL } |
| | 786 | }; |
| | 787 | |
| | 788 | AVFILTER_DEFINE_CLASS(headphone); |
| | 789 | |
| | 790 | static const AVFilterPad outputs[] = { |
| | 791 | { |
| | 792 | .name = "default", |
| | 793 | .type = AVMEDIA_TYPE_AUDIO, |
| | 794 | .config_props = config_output, |
| | 795 | .request_frame = request_frame, |
| | 796 | }, |
| | 797 | { NULL } |
| | 798 | }; |
| | 799 | |
| | 800 | AVFilter ff_af_headphone = { |
| | 801 | .name = "headphone", |
| | 802 | .description = NULL_IF_CONFIG_SMALL("Apply headphone binaural spatialization with HRTFs in additional streams."), |
| | 803 | .priv_size = sizeof(HeadphoneContext), |
| | 804 | .priv_class = &headphone_class, |
| | 805 | .init = init, |
| | 806 | .uninit = uninit, |
| | 807 | .query_formats = query_formats, |
| | 808 | .inputs = NULL, |
| | 809 | .outputs = outputs, |
| | 810 | .flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS, |
| | 811 | }; |