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  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. #include <math.h>
  21. #include "libavutil/avstring.h"
  22. #include "libavutil/channel_layout.h"
  23. #include "libavutil/float_dsp.h"
  24. #include "libavutil/intmath.h"
  25. #include "libavutil/opt.h"
  26. #include "libavcodec/avfft.h"
  27. #include "avfilter.h"
  28. #include "filters.h"
  29. #include "internal.h"
  30. #include "audio.h"
  31. #define TIME_DOMAIN 0
  32. #define FREQUENCY_DOMAIN 1
  33. #define HRIR_STEREO 0
  34. #define HRIR_MULTI 1
  35. typedef struct HeadphoneContext {
  36. const AVClass *class;
  37. char *map;
  38. int type;
  39. int lfe_channel;
  40. int have_hrirs;
  41. int eof_hrirs;
  42. int ir_len;
  43. int mapping[64];
  44. int nb_inputs;
  45. int nb_irs;
  46. float gain;
  47. float lfe_gain, gain_lfe;
  48. float *ringbuffer[2];
  49. int write[2];
  50. int buffer_length;
  51. int n_fft;
  52. int size;
  53. int hrir_fmt;
  54. int *delay[2];
  55. float *data_ir[2];
  56. float *temp_src[2];
  57. FFTComplex *temp_fft[2];
  58. FFTContext *fft[2], *ifft[2];
  59. FFTComplex *data_hrtf[2];
  60. AVFloatDSPContext *fdsp;
  61. struct headphone_inputs {
  62. AVFrame *frame;
  63. int ir_len;
  64. int delay_l;
  65. int delay_r;
  66. int eof;
  67. } *in;
  68. } HeadphoneContext;
  69. static int parse_channel_name(HeadphoneContext *s, int x, char **arg, int *rchannel, char *buf)
  70. {
  71. int len, i, channel_id = 0;
  72. int64_t layout, layout0;
  73. if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) {
  74. layout0 = layout = av_get_channel_layout(buf);
  75. if (layout == AV_CH_LOW_FREQUENCY)
  76. s->lfe_channel = x;
  77. for (i = 32; i > 0; i >>= 1) {
  78. if (layout >= 1LL << i) {
  79. channel_id += i;
  80. layout >>= i;
  81. }
  82. }
  83. if (channel_id >= 64 || layout0 != 1LL << channel_id)
  84. return AVERROR(EINVAL);
  85. *rchannel = channel_id;
  86. *arg += len;
  87. return 0;
  88. }
  89. return AVERROR(EINVAL);
  90. }
  91. static void parse_map(AVFilterContext *ctx)
  92. {
  93. HeadphoneContext *s = ctx->priv;
  94. char *arg, *tokenizer, *p, *args = av_strdup(s->map);
  95. int i;
  96. if (!args)
  97. return;
  98. p = args;
  99. s->lfe_channel = -1;
  100. s->nb_inputs = 1;
  101. for (i = 0; i < 64; i++) {
  102. s->mapping[i] = -1;
  103. }
  104. while ((arg = av_strtok(p, "|", &tokenizer))) {
  105. int out_ch_id;
  106. char buf[8];
  107. p = NULL;
  108. if (parse_channel_name(s, s->nb_irs, &arg, &out_ch_id, buf)) {
  109. av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf);
  110. continue;
  111. }
  112. s->mapping[s->nb_irs] = out_ch_id;
  113. s->nb_irs++;
  114. }
  115. if (s->hrir_fmt == HRIR_MULTI)
  116. s->nb_inputs = 2;
  117. else
  118. s->nb_inputs = s->nb_irs + 1;
  119. av_free(args);
  120. }
  121. typedef struct ThreadData {
  122. AVFrame *in, *out;
  123. int *write;
  124. int **delay;
  125. float **ir;
  126. int *n_clippings;
  127. float **ringbuffer;
  128. float **temp_src;
  129. FFTComplex **temp_fft;
  130. } ThreadData;
  131. static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
  132. {
  133. HeadphoneContext *s = ctx->priv;
  134. ThreadData *td = arg;
  135. AVFrame *in = td->in, *out = td->out;
  136. int offset = jobnr;
  137. int *write = &td->write[jobnr];
  138. const int *const delay = td->delay[jobnr];
  139. const float *const ir = td->ir[jobnr];
  140. int *n_clippings = &td->n_clippings[jobnr];
  141. float *ringbuffer = td->ringbuffer[jobnr];
  142. float *temp_src = td->temp_src[jobnr];
  143. const int ir_len = s->ir_len;
  144. const float *src = (const float *)in->data[0];
  145. float *dst = (float *)out->data[0];
  146. const int in_channels = in->channels;
  147. const int buffer_length = s->buffer_length;
  148. const uint32_t modulo = (uint32_t)buffer_length - 1;
  149. float *buffer[16];
  150. int wr = *write;
  151. int read;
  152. int i, l;
  153. dst += offset;
  154. for (l = 0; l < in_channels; l++) {
  155. buffer[l] = ringbuffer + l * buffer_length;
  156. }
  157. for (i = 0; i < in->nb_samples; i++) {
  158. const float *temp_ir = ir;
  159. *dst = 0;
  160. for (l = 0; l < in_channels; l++) {
  161. *(buffer[l] + wr) = src[l];
  162. }
  163. for (l = 0; l < in_channels; l++) {
  164. const float *const bptr = buffer[l];
  165. if (l == s->lfe_channel) {
  166. *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
  167. temp_ir += FFALIGN(ir_len, 16);
  168. continue;
  169. }
  170. read = (wr - *(delay + l) - (ir_len - 1) + buffer_length) & modulo;
  171. if (read + ir_len < buffer_length) {
  172. memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
  173. } else {
  174. int len = FFMIN(ir_len - (read % ir_len), buffer_length - read);
  175. memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
  176. memcpy(temp_src + len, bptr, (ir_len - len) * sizeof(*temp_src));
  177. }
  178. dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, ir_len);
  179. temp_ir += FFALIGN(ir_len, 16);
  180. }
  181. if (fabs(*dst) > 1)
  182. *n_clippings += 1;
  183. dst += 2;
  184. src += in_channels;
  185. wr = (wr + 1) & modulo;
  186. }
  187. *write = wr;
  188. return 0;
  189. }
  190. static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
  191. {
  192. HeadphoneContext *s = ctx->priv;
  193. ThreadData *td = arg;
  194. AVFrame *in = td->in, *out = td->out;
  195. int offset = jobnr;
  196. int *write = &td->write[jobnr];
  197. FFTComplex *hrtf = s->data_hrtf[jobnr];
  198. int *n_clippings = &td->n_clippings[jobnr];
  199. float *ringbuffer = td->ringbuffer[jobnr];
  200. const int ir_len = s->ir_len;
  201. const float *src = (const float *)in->data[0];
  202. float *dst = (float *)out->data[0];
  203. const int in_channels = in->channels;
  204. const int buffer_length = s->buffer_length;
  205. const uint32_t modulo = (uint32_t)buffer_length - 1;
  206. FFTComplex *fft_in = s->temp_fft[jobnr];
  207. FFTContext *ifft = s->ifft[jobnr];
  208. FFTContext *fft = s->fft[jobnr];
  209. const int n_fft = s->n_fft;
  210. const float fft_scale = 1.0f / s->n_fft;
  211. FFTComplex *hrtf_offset;
  212. int wr = *write;
  213. int n_read;
  214. int i, j;
  215. dst += offset;
  216. n_read = FFMIN(s->ir_len, in->nb_samples);
  217. for (j = 0; j < n_read; j++) {
  218. dst[2 * j] = ringbuffer[wr];
  219. ringbuffer[wr] = 0.0;
  220. wr = (wr + 1) & modulo;
  221. }
  222. for (j = n_read; j < in->nb_samples; j++) {
  223. dst[2 * j] = 0;
  224. }
  225. for (i = 0; i < in_channels; i++) {
  226. if (i == s->lfe_channel) {
  227. for (j = 0; j < in->nb_samples; j++) {
  228. dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
  229. }
  230. continue;
  231. }
  232. offset = i * n_fft;
  233. hrtf_offset = hrtf + offset;
  234. memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
  235. for (j = 0; j < in->nb_samples; j++) {
  236. fft_in[j].re = src[j * in_channels + i];
  237. }
  238. av_fft_permute(fft, fft_in);
  239. av_fft_calc(fft, fft_in);
  240. for (j = 0; j < n_fft; j++) {
  241. const FFTComplex *hcomplex = hrtf_offset + j;
  242. const float re = fft_in[j].re;
  243. const float im = fft_in[j].im;
  244. fft_in[j].re = re * hcomplex->re - im * hcomplex->im;
  245. fft_in[j].im = re * hcomplex->im + im * hcomplex->re;
  246. }
  247. av_fft_permute(ifft, fft_in);
  248. av_fft_calc(ifft, fft_in);
  249. for (j = 0; j < in->nb_samples; j++) {
  250. dst[2 * j] += fft_in[j].re * fft_scale;
  251. }
  252. for (j = 0; j < ir_len - 1; j++) {
  253. int write_pos = (wr + j) & modulo;
  254. *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale;
  255. }
  256. }
  257. for (i = 0; i < out->nb_samples; i++) {
  258. if (fabs(*dst) > 1) {
  259. n_clippings[0]++;
  260. }
  261. dst += 2;
  262. }
  263. *write = wr;
  264. return 0;
  265. }
  266. static int check_ir(AVFilterLink *inlink, int input_number)
  267. {
  268. AVFilterContext *ctx = inlink->dst;
  269. HeadphoneContext *s = ctx->priv;
  270. int ir_len, max_ir_len;
  271. ir_len = ff_inlink_queued_samples(inlink);
  272. max_ir_len = 65536;
  273. if (ir_len > max_ir_len) {
  274. av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len);
  275. return AVERROR(EINVAL);
  276. }
  277. s->in[input_number].ir_len = ir_len;
  278. s->ir_len = FFMAX(ir_len, s->ir_len);
  279. return 0;
  280. }
  281. static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink)
  282. {
  283. AVFilterContext *ctx = outlink->src;
  284. int n_clippings[2] = { 0 };
  285. ThreadData td;
  286. AVFrame *out;
  287. out = ff_get_audio_buffer(outlink, in->nb_samples);
  288. if (!out) {
  289. av_frame_free(&in);
  290. return AVERROR(ENOMEM);
  291. }
  292. out->pts = in->pts;
  293. td.in = in; td.out = out; td.write = s->write;
  294. td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
  295. td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
  296. td.temp_fft = s->temp_fft;
  297. if (s->type == TIME_DOMAIN) {
  298. ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2);
  299. } else {
  300. ctx->internal->execute(ctx, headphone_fast_convolute, &td, NULL, 2);
  301. }
  302. emms_c();
  303. if (n_clippings[0] + n_clippings[1] > 0) {
  304. av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",
  305. n_clippings[0] + n_clippings[1], out->nb_samples * 2);
  306. }
  307. av_frame_free(&in);
  308. return ff_filter_frame(outlink, out);
  309. }
  310. static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
  311. {
  312. struct HeadphoneContext *s = ctx->priv;
  313. const int ir_len = s->ir_len;
  314. int nb_irs = s->nb_irs;
  315. int nb_input_channels = ctx->inputs[0]->channels;
  316. float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
  317. FFTComplex *data_hrtf_l = NULL;
  318. FFTComplex *data_hrtf_r = NULL;
  319. FFTComplex *fft_in_l = NULL;
  320. FFTComplex *fft_in_r = NULL;
  321. float *data_ir_l = NULL;
  322. float *data_ir_r = NULL;
  323. int offset = 0, ret = 0;
  324. int n_fft;
  325. int i, j, k;
  326. s->buffer_length = 1 << (32 - ff_clz(s->ir_len));
  327. s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + s->size));
  328. if (s->type == FREQUENCY_DOMAIN) {
  329. fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
  330. fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
  331. if (!fft_in_l || !fft_in_r) {
  332. ret = AVERROR(ENOMEM);
  333. goto fail;
  334. }
  335. av_fft_end(s->fft[0]);
  336. av_fft_end(s->fft[1]);
  337. s->fft[0] = av_fft_init(log2(s->n_fft), 0);
  338. s->fft[1] = av_fft_init(log2(s->n_fft), 0);
  339. av_fft_end(s->ifft[0]);
  340. av_fft_end(s->ifft[1]);
  341. s->ifft[0] = av_fft_init(log2(s->n_fft), 1);
  342. s->ifft[1] = av_fft_init(log2(s->n_fft), 1);
  343. if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {
  344. av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft);
  345. ret = AVERROR(ENOMEM);
  346. goto fail;
  347. }
  348. }
  349. s->data_ir[0] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
  350. s->data_ir[1] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
  351. s->delay[0] = av_calloc(s->nb_irs, sizeof(float));
  352. s->delay[1] = av_calloc(s->nb_irs, sizeof(float));
  353. if (s->type == TIME_DOMAIN) {
  354. s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
  355. s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
  356. } else {
  357. s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
  358. s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
  359. s->temp_fft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
  360. s->temp_fft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
  361. if (!s->temp_fft[0] || !s->temp_fft[1]) {
  362. ret = AVERROR(ENOMEM);
  363. goto fail;
  364. }
  365. }
  366. if (!s->data_ir[0] || !s->data_ir[1] ||
  367. !s->ringbuffer[0] || !s->ringbuffer[1]) {
  368. ret = AVERROR(ENOMEM);
  369. goto fail;
  370. }
  371. if (s->type == TIME_DOMAIN) {
  372. s->temp_src[0] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
  373. s->temp_src[1] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
  374. data_ir_l = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_l));
  375. data_ir_r = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_r));
  376. if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) {
  377. ret = AVERROR(ENOMEM);
  378. goto fail;
  379. }
  380. } else {
  381. data_hrtf_l = av_calloc(n_fft, sizeof(*data_hrtf_l) * nb_irs);
  382. data_hrtf_r = av_calloc(n_fft, sizeof(*data_hrtf_r) * nb_irs);
  383. if (!data_hrtf_r || !data_hrtf_l) {
  384. ret = AVERROR(ENOMEM);
  385. goto fail;
  386. }
  387. }
  388. for (i = 0; i < s->nb_inputs - 1; i++) {
  389. int len = s->in[i + 1].ir_len;
  390. int delay_l = s->in[i + 1].delay_l;
  391. int delay_r = s->in[i + 1].delay_r;
  392. float *ptr;
  393. ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &s->in[i + 1].frame);
  394. if (ret < 0)
  395. goto fail;
  396. ptr = (float *)s->in[i + 1].frame->extended_data[0];
  397. if (s->hrir_fmt == HRIR_STEREO) {
  398. int idx = -1;
  399. for (j = 0; j < inlink->channels; j++) {
  400. if (s->mapping[i] < 0) {
  401. continue;
  402. }
  403. if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[i])) {
  404. idx = i;
  405. break;
  406. }
  407. }
  408. if (idx == -1)
  409. continue;
  410. if (s->type == TIME_DOMAIN) {
  411. offset = idx * FFALIGN(len, 16);
  412. for (j = 0; j < len; j++) {
  413. data_ir_l[offset + j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
  414. data_ir_r[offset + j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
  415. }
  416. } else {
  417. memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
  418. memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
  419. offset = idx * n_fft;
  420. for (j = 0; j < len; j++) {
  421. fft_in_l[delay_l + j].re = ptr[j * 2 ] * gain_lin;
  422. fft_in_r[delay_r + j].re = ptr[j * 2 + 1] * gain_lin;
  423. }
  424. av_fft_permute(s->fft[0], fft_in_l);
  425. av_fft_calc(s->fft[0], fft_in_l);
  426. memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
  427. av_fft_permute(s->fft[0], fft_in_r);
  428. av_fft_calc(s->fft[0], fft_in_r);
  429. memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
  430. }
  431. } else {
  432. int I, N = ctx->inputs[1]->channels;
  433. for (k = 0; k < N / 2; k++) {
  434. int idx = -1;
  435. for (j = 0; j < inlink->channels; j++) {
  436. if (s->mapping[k] < 0) {
  437. continue;
  438. }
  439. if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[k])) {
  440. idx = k;
  441. break;
  442. }
  443. }
  444. if (idx == -1)
  445. continue;
  446. I = idx * 2;
  447. if (s->type == TIME_DOMAIN) {
  448. offset = idx * FFALIGN(len, 16);
  449. for (j = 0; j < len; j++) {
  450. data_ir_l[offset + j] = ptr[len * N - j * N - N + I ] * gain_lin;
  451. data_ir_r[offset + j] = ptr[len * N - j * N - N + I + 1] * gain_lin;
  452. }
  453. } else {
  454. memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
  455. memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
  456. offset = idx * n_fft;
  457. for (j = 0; j < len; j++) {
  458. fft_in_l[delay_l + j].re = ptr[j * N + I ] * gain_lin;
  459. fft_in_r[delay_r + j].re = ptr[j * N + I + 1] * gain_lin;
  460. }
  461. av_fft_permute(s->fft[0], fft_in_l);
  462. av_fft_calc(s->fft[0], fft_in_l);
  463. memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
  464. av_fft_permute(s->fft[0], fft_in_r);
  465. av_fft_calc(s->fft[0], fft_in_r);
  466. memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
  467. }
  468. }
  469. }
  470. av_frame_free(&s->in[i + 1].frame);
  471. }
  472. if (s->type == TIME_DOMAIN) {
  473. memcpy(s->data_ir[0], data_ir_l, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
  474. memcpy(s->data_ir[1], data_ir_r, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
  475. } else {
  476. s->data_hrtf[0] = av_calloc(n_fft * s->nb_irs, sizeof(FFTComplex));
  477. s->data_hrtf[1] = av_calloc(n_fft * s->nb_irs, sizeof(FFTComplex));
  478. if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
  479. ret = AVERROR(ENOMEM);
  480. goto fail;
  481. }
  482. memcpy(s->data_hrtf[0], data_hrtf_l,
  483. sizeof(FFTComplex) * nb_irs * n_fft);
  484. memcpy(s->data_hrtf[1], data_hrtf_r,
  485. sizeof(FFTComplex) * nb_irs * n_fft);
  486. }
  487. s->have_hrirs = 1;
  488. fail:
  489. for (i = 0; i < s->nb_inputs - 1; i++)
  490. av_frame_free(&s->in[i + 1].frame);
  491. av_freep(&data_ir_l);
  492. av_freep(&data_ir_r);
  493. av_freep(&data_hrtf_l);
  494. av_freep(&data_hrtf_r);
  495. av_freep(&fft_in_l);
  496. av_freep(&fft_in_r);
  497. return ret;
  498. }
  499. static int activate(AVFilterContext *ctx)
  500. {
  501. HeadphoneContext *s = ctx->priv;
  502. AVFilterLink *inlink = ctx->inputs[0];
  503. AVFilterLink *outlink = ctx->outputs[0];
  504. AVFrame *in = NULL;
  505. int i, ret;
  506. FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
  507. if (!s->eof_hrirs) {
  508. for (i = 1; i < s->nb_inputs; i++) {
  509. if (s->in[i].eof)
  510. continue;
  511. if ((ret = check_ir(ctx->inputs[i], i)) < 0)
  512. return ret;
  513. if (!s->in[i].eof) {
  514. if (ff_outlink_get_status(ctx->inputs[i]) == AVERROR_EOF)
  515. s->in[i].eof = 1;
  516. }
  517. }
  518. for (i = 1; i < s->nb_inputs; i++) {
  519. if (!s->in[i].eof)
  520. break;
  521. }
  522. if (i != s->nb_inputs) {
  523. if (ff_outlink_frame_wanted(ctx->outputs[0])) {
  524. for (i = 1; i < s->nb_inputs; i++) {
  525. if (!s->in[i].eof)
  526. ff_inlink_request_frame(ctx->inputs[i]);
  527. }
  528. }
  529. return 0;
  530. } else {
  531. s->eof_hrirs = 1;
  532. }
  533. }
  534. if (!s->have_hrirs && s->eof_hrirs) {
  535. ret = convert_coeffs(ctx, inlink);
  536. if (ret < 0)
  537. return ret;
  538. }
  539. if ((ret = ff_inlink_consume_samples(ctx->inputs[0], s->size, s->size, &in)) > 0) {
  540. ret = headphone_frame(s, in, outlink);
  541. if (ret < 0)
  542. return ret;
  543. }
  544. if (ret < 0)
  545. return ret;
  546. FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
  547. if (ff_outlink_frame_wanted(ctx->outputs[0]))
  548. ff_inlink_request_frame(ctx->inputs[0]);
  549. return 0;
  550. }
  551. static int query_formats(AVFilterContext *ctx)
  552. {
  553. struct HeadphoneContext *s = ctx->priv;
  554. AVFilterFormats *formats = NULL;
  555. AVFilterChannelLayouts *layouts = NULL;
  556. AVFilterChannelLayouts *stereo_layout = NULL;
  557. AVFilterChannelLayouts *hrir_layouts = NULL;
  558. int ret, i;
  559. ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
  560. if (ret)
  561. return ret;
  562. ret = ff_set_common_formats(ctx, formats);
  563. if (ret)
  564. return ret;
  565. layouts = ff_all_channel_layouts();
  566. if (!layouts)
  567. return AVERROR(ENOMEM);
  568. ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts);
  569. if (ret)
  570. return ret;
  571. ret = ff_add_channel_layout(&stereo_layout, AV_CH_LAYOUT_STEREO);
  572. if (ret)
  573. return ret;
  574. if (s->hrir_fmt == HRIR_MULTI) {
  575. hrir_layouts = ff_all_channel_counts();
  576. if (!hrir_layouts)
  577. ret = AVERROR(ENOMEM);
  578. ret = ff_channel_layouts_ref(hrir_layouts, &ctx->inputs[1]->out_channel_layouts);
  579. if (ret)
  580. return ret;
  581. } else {
  582. for (i = 1; i < s->nb_inputs; i++) {
  583. ret = ff_channel_layouts_ref(stereo_layout, &ctx->inputs[i]->out_channel_layouts);
  584. if (ret)
  585. return ret;
  586. }
  587. }
  588. ret = ff_channel_layouts_ref(stereo_layout, &ctx->outputs[0]->in_channel_layouts);
  589. if (ret)
  590. return ret;
  591. formats = ff_all_samplerates();
  592. if (!formats)
  593. return AVERROR(ENOMEM);
  594. return ff_set_common_samplerates(ctx, formats);
  595. }
  596. static int config_input(AVFilterLink *inlink)
  597. {
  598. AVFilterContext *ctx = inlink->dst;
  599. HeadphoneContext *s = ctx->priv;
  600. if (s->nb_irs < inlink->channels) {
  601. av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->channels);
  602. return AVERROR(EINVAL);
  603. }
  604. return 0;
  605. }
  606. static av_cold int init(AVFilterContext *ctx)
  607. {
  608. HeadphoneContext *s = ctx->priv;
  609. int i, ret;
  610. AVFilterPad pad = {
  611. .name = "in0",
  612. .type = AVMEDIA_TYPE_AUDIO,
  613. .config_props = config_input,
  614. };
  615. if ((ret = ff_insert_inpad(ctx, 0, &pad)) < 0)
  616. return ret;
  617. if (!s->map) {
  618. av_log(ctx, AV_LOG_ERROR, "Valid mapping must be set.\n");
  619. return AVERROR(EINVAL);
  620. }
  621. parse_map(ctx);
  622. s->in = av_calloc(s->nb_inputs, sizeof(*s->in));
  623. if (!s->in)
  624. return AVERROR(ENOMEM);
  625. for (i = 1; i < s->nb_inputs; i++) {
  626. char *name = av_asprintf("hrir%d", i - 1);
  627. AVFilterPad pad = {
  628. .name = name,
  629. .type = AVMEDIA_TYPE_AUDIO,
  630. };
  631. if (!name)
  632. return AVERROR(ENOMEM);
  633. if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
  634. av_freep(&pad.name);
  635. return ret;
  636. }
  637. }
  638. s->fdsp = avpriv_float_dsp_alloc(0);
  639. if (!s->fdsp)
  640. return AVERROR(ENOMEM);
  641. return 0;
  642. }
  643. static int config_output(AVFilterLink *outlink)
  644. {
  645. AVFilterContext *ctx = outlink->src;
  646. HeadphoneContext *s = ctx->priv;
  647. AVFilterLink *inlink = ctx->inputs[0];
  648. if (s->hrir_fmt == HRIR_MULTI) {
  649. AVFilterLink *hrir_link = ctx->inputs[1];
  650. if (hrir_link->channels < inlink->channels * 2) {
  651. av_log(ctx, AV_LOG_ERROR, "Number of channels in HRIR stream must be >= %d.\n", inlink->channels * 2);
  652. return AVERROR(EINVAL);
  653. }
  654. }
  655. s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6 + s->lfe_gain) / 20 * M_LN10);
  656. return 0;
  657. }
  658. static av_cold void uninit(AVFilterContext *ctx)
  659. {
  660. HeadphoneContext *s = ctx->priv;
  661. int i;
  662. av_fft_end(s->ifft[0]);
  663. av_fft_end(s->ifft[1]);
  664. av_fft_end(s->fft[0]);
  665. av_fft_end(s->fft[1]);
  666. av_freep(&s->delay[0]);
  667. av_freep(&s->delay[1]);
  668. av_freep(&s->data_ir[0]);
  669. av_freep(&s->data_ir[1]);
  670. av_freep(&s->ringbuffer[0]);
  671. av_freep(&s->ringbuffer[1]);
  672. av_freep(&s->temp_src[0]);
  673. av_freep(&s->temp_src[1]);
  674. av_freep(&s->temp_fft[0]);
  675. av_freep(&s->temp_fft[1]);
  676. av_freep(&s->data_hrtf[0]);
  677. av_freep(&s->data_hrtf[1]);
  678. av_freep(&s->fdsp);
  679. for (i = 0; i < s->nb_inputs; i++) {
  680. if (ctx->input_pads && i)
  681. av_freep(&ctx->input_pads[i].name);
  682. }
  683. av_freep(&s->in);
  684. }
  685. #define OFFSET(x) offsetof(HeadphoneContext, x)
  686. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  687. static const AVOption headphone_options[] = {
  688. { "map", "set channels convolution mappings", OFFSET(map), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
  689. { "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS },
  690. { "lfe", "set lfe gain in dB", OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS },
  691. { "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" },
  692. { "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" },
  693. { "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" },
  694. { "size", "set frame size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=1024},1024,96000, .flags = FLAGS },
  695. { "hrir", "set hrir format", OFFSET(hrir_fmt), AV_OPT_TYPE_INT, {.i64=HRIR_STEREO}, 0, 1, .flags = FLAGS, "hrir" },
  696. { "stereo", "hrir files have exactly 2 channels", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_STEREO}, 0, 0, .flags = FLAGS, "hrir" },
  697. { "multich", "single multichannel hrir file", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_MULTI}, 0, 0, .flags = FLAGS, "hrir" },
  698. { NULL }
  699. };
  700. AVFILTER_DEFINE_CLASS(headphone);
  701. static const AVFilterPad outputs[] = {
  702. {
  703. .name = "default",
  704. .type = AVMEDIA_TYPE_AUDIO,
  705. .config_props = config_output,
  706. },
  707. { NULL }
  708. };
  709. AVFilter ff_af_headphone = {
  710. .name = "headphone",
  711. .description = NULL_IF_CONFIG_SMALL("Apply headphone binaural spatialization with HRTFs in additional streams."),
  712. .priv_size = sizeof(HeadphoneContext),
  713. .priv_class = &headphone_class,
  714. .init = init,
  715. .uninit = uninit,
  716. .query_formats = query_formats,
  717. .activate = activate,
  718. .inputs = NULL,
  719. .outputs = outputs,
  720. .flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
  721. };