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  1. /*
  2. * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
  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. #include "libavutil/channel_layout.h"
  21. #include "libavutil/opt.h"
  22. #include "avfilter.h"
  23. #include "audio.h"
  24. #include "formats.h"
  25. typedef struct StereoToolsContext {
  26. const AVClass *class;
  27. int softclip;
  28. int mute_l;
  29. int mute_r;
  30. int phase_l;
  31. int phase_r;
  32. int mode;
  33. int bmode_in;
  34. int bmode_out;
  35. double slev;
  36. double sbal;
  37. double mlev;
  38. double mpan;
  39. double phase;
  40. double base;
  41. double delay;
  42. double balance_in;
  43. double balance_out;
  44. double phase_sin_coef;
  45. double phase_cos_coef;
  46. double sc_level;
  47. double inv_atan_shape;
  48. double level_in;
  49. double level_out;
  50. double *buffer;
  51. int length;
  52. int pos;
  53. } StereoToolsContext;
  54. #define OFFSET(x) offsetof(StereoToolsContext, x)
  55. #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  56. static const AVOption stereotools_options[] = {
  57. { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
  58. { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
  59. { "balance_in", "set balance in", OFFSET(balance_in), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
  60. { "balance_out", "set balance out", OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
  61. { "softclip", "enable softclip", OFFSET(softclip), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
  62. { "mutel", "mute L", OFFSET(mute_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
  63. { "muter", "mute R", OFFSET(mute_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
  64. { "phasel", "phase L", OFFSET(phase_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
  65. { "phaser", "phase R", OFFSET(phase_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
  66. { "mode", "set stereo mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 6, A, "mode" },
  67. { "lr>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
  68. { "lr>ms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
  69. { "ms>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "mode" },
  70. { "lr>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "mode" },
  71. { "lr>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, A, "mode" },
  72. { "lr>l+r", 0, 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, A, "mode" },
  73. { "lr>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, A, "mode" },
  74. { "slev", "set side level", OFFSET(slev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
  75. { "sbal", "set side balance", OFFSET(sbal), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
  76. { "mlev", "set middle level", OFFSET(mlev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
  77. { "mpan", "set middle pan", OFFSET(mpan), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
  78. { "base", "set stereo base", OFFSET(base), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
  79. { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20, 20, A },
  80. { "sclevel", "set S/C level", OFFSET(sc_level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 100, A },
  81. { "phase", "set stereo phase", OFFSET(phase), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 360, A },
  82. { "bmode_in", "set balance in mode", OFFSET(bmode_in), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
  83. { "balance", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "bmode" },
  84. { "amplitude", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "bmode" },
  85. { "power", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "bmode" },
  86. { "bmode_out", "set balance out mode", OFFSET(bmode_out), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
  87. { NULL }
  88. };
  89. AVFILTER_DEFINE_CLASS(stereotools);
  90. static int query_formats(AVFilterContext *ctx)
  91. {
  92. AVFilterFormats *formats = NULL;
  93. AVFilterChannelLayouts *layout = NULL;
  94. int ret;
  95. if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
  96. (ret = ff_set_common_formats (ctx , formats )) < 0 ||
  97. (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
  98. (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
  99. return ret;
  100. formats = ff_all_samplerates();
  101. return ff_set_common_samplerates(ctx, formats);
  102. }
  103. static int config_input(AVFilterLink *inlink)
  104. {
  105. AVFilterContext *ctx = inlink->dst;
  106. StereoToolsContext *s = ctx->priv;
  107. s->length = 2 * inlink->sample_rate * 0.05;
  108. if (s->length <= 1 || s->length & 1) {
  109. av_log(ctx, AV_LOG_ERROR, "sample rate is too small\n");
  110. return AVERROR(EINVAL);
  111. }
  112. s->buffer = av_calloc(s->length, sizeof(*s->buffer));
  113. if (!s->buffer)
  114. return AVERROR(ENOMEM);
  115. s->inv_atan_shape = 1.0 / atan(s->sc_level);
  116. s->phase_cos_coef = cos(s->phase / 180 * M_PI);
  117. s->phase_sin_coef = sin(s->phase / 180 * M_PI);
  118. return 0;
  119. }
  120. static int filter_frame(AVFilterLink *inlink, AVFrame *in)
  121. {
  122. AVFilterContext *ctx = inlink->dst;
  123. AVFilterLink *outlink = ctx->outputs[0];
  124. StereoToolsContext *s = ctx->priv;
  125. const double *src = (const double *)in->data[0];
  126. const double sb = s->base < 0 ? s->base * 0.5 : s->base;
  127. const double sbal = 1 + s->sbal;
  128. const double mpan = 1 + s->mpan;
  129. const double slev = s->slev;
  130. const double mlev = s->mlev;
  131. const double balance_in = s->balance_in;
  132. const double balance_out = s->balance_out;
  133. const double level_in = s->level_in;
  134. const double level_out = s->level_out;
  135. const double sc_level = s->sc_level;
  136. const double delay = s->delay;
  137. const int length = s->length;
  138. const int mute_l = s->mute_l;
  139. const int mute_r = s->mute_r;
  140. const int phase_l = s->phase_l;
  141. const int phase_r = s->phase_r;
  142. double *buffer = s->buffer;
  143. AVFrame *out;
  144. double *dst;
  145. int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
  146. int n;
  147. nbuf -= nbuf % 2;
  148. if (av_frame_is_writable(in)) {
  149. out = in;
  150. } else {
  151. out = ff_get_audio_buffer(inlink, in->nb_samples);
  152. if (!out) {
  153. av_frame_free(&in);
  154. return AVERROR(ENOMEM);
  155. }
  156. av_frame_copy_props(out, in);
  157. }
  158. dst = (double *)out->data[0];
  159. for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
  160. double L = src[0], R = src[1], l, r, m, S, gl, gr, gd;
  161. L *= level_in;
  162. R *= level_in;
  163. gl = 1. - FFMAX(0., balance_in);
  164. gr = 1. + FFMIN(0., balance_in);
  165. switch (s->bmode_in) {
  166. case 1:
  167. gd = gl - gr;
  168. gl = 1. + gd;
  169. gr = 1. - gd;
  170. break;
  171. case 2:
  172. if (balance_in < 0.) {
  173. gr = FFMAX(0.5, gr);
  174. gl = 1. / gr;
  175. } else if (balance_in > 0.) {
  176. gl = FFMAX(0.5, gl);
  177. gr = 1. / gl;
  178. }
  179. break;
  180. }
  181. L *= gl;
  182. R *= gr;
  183. if (s->softclip) {
  184. R = s->inv_atan_shape * atan(R * sc_level);
  185. L = s->inv_atan_shape * atan(L * sc_level);
  186. }
  187. switch (s->mode) {
  188. case 0:
  189. m = (L + R) * 0.5;
  190. S = (L - R) * 0.5;
  191. l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
  192. r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
  193. L = l;
  194. R = r;
  195. break;
  196. case 1:
  197. l = L * FFMIN(1., 2. - sbal);
  198. r = R * FFMIN(1., sbal);
  199. L = 0.5 * (l + r) * mlev;
  200. R = 0.5 * (l - r) * slev;
  201. break;
  202. case 2:
  203. l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
  204. r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
  205. L = l;
  206. R = r;
  207. break;
  208. case 3:
  209. R = L;
  210. break;
  211. case 4:
  212. L = R;
  213. break;
  214. case 5:
  215. L = (L + R) / 2;
  216. R = L;
  217. break;
  218. case 6:
  219. l = L;
  220. L = R;
  221. R = l;
  222. m = (L + R) * 0.5;
  223. S = (L - R) * 0.5;
  224. l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
  225. r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
  226. L = l;
  227. R = r;
  228. break;
  229. }
  230. L *= 1. - mute_l;
  231. R *= 1. - mute_r;
  232. L *= (2. * (1. - phase_l)) - 1.;
  233. R *= (2. * (1. - phase_r)) - 1.;
  234. buffer[s->pos ] = L;
  235. buffer[s->pos+1] = R;
  236. if (delay > 0.) {
  237. R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
  238. } else if (delay < 0.) {
  239. L = buffer[(s->pos - (int)nbuf + length) % length];
  240. }
  241. l = L + sb * L - sb * R;
  242. r = R + sb * R - sb * L;
  243. L = l;
  244. R = r;
  245. l = L * s->phase_cos_coef - R * s->phase_sin_coef;
  246. r = L * s->phase_sin_coef + R * s->phase_cos_coef;
  247. L = l;
  248. R = r;
  249. s->pos = (s->pos + 2) % s->length;
  250. gl = 1. - FFMAX(0., balance_out);
  251. gr = 1. + FFMIN(0., balance_out);
  252. switch (s->bmode_out) {
  253. case 1:
  254. gd = gl - gr;
  255. gl = 1. + gd;
  256. gr = 1. - gd;
  257. break;
  258. case 2:
  259. if (balance_out < 0.) {
  260. gr = FFMAX(0.5, gr);
  261. gl = 1. / gr;
  262. } else if (balance_out > 0.) {
  263. gl = FFMAX(0.5, gl);
  264. gr = 1. / gl;
  265. }
  266. break;
  267. }
  268. L *= gl;
  269. R *= gr;
  270. L *= level_out;
  271. R *= level_out;
  272. dst[0] = L;
  273. dst[1] = R;
  274. }
  275. if (out != in)
  276. av_frame_free(&in);
  277. return ff_filter_frame(outlink, out);
  278. }
  279. static av_cold void uninit(AVFilterContext *ctx)
  280. {
  281. StereoToolsContext *s = ctx->priv;
  282. av_freep(&s->buffer);
  283. }
  284. static const AVFilterPad inputs[] = {
  285. {
  286. .name = "default",
  287. .type = AVMEDIA_TYPE_AUDIO,
  288. .filter_frame = filter_frame,
  289. .config_props = config_input,
  290. },
  291. { NULL }
  292. };
  293. static const AVFilterPad outputs[] = {
  294. {
  295. .name = "default",
  296. .type = AVMEDIA_TYPE_AUDIO,
  297. },
  298. { NULL }
  299. };
  300. AVFilter ff_af_stereotools = {
  301. .name = "stereotools",
  302. .description = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
  303. .query_formats = query_formats,
  304. .priv_size = sizeof(StereoToolsContext),
  305. .priv_class = &stereotools_class,
  306. .uninit = uninit,
  307. .inputs = inputs,
  308. .outputs = outputs,
  309. };