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