af_aresample.c 11 KB

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  1. /*
  2. * Copyright (c) 2011 Stefano Sabatini
  3. * Copyright (c) 2011 Mina Nagy Zaki
  4. *
  5. * This file is part of FFmpeg.
  6. *
  7. * FFmpeg is free software; you can redistribute it and/or
  8. * modify it under the terms of the GNU Lesser General Public
  9. * License as published by the Free Software Foundation; either
  10. * version 2.1 of the License, or (at your option) any later version.
  11. *
  12. * FFmpeg is distributed in the hope that it will be useful,
  13. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  15. * Lesser General Public License for more details.
  16. *
  17. * You should have received a copy of the GNU Lesser General Public
  18. * License along with FFmpeg; if not, write to the Free Software
  19. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20. */
  21. /**
  22. * @file
  23. * resampling audio filter
  24. */
  25. #include "libavutil/eval.h"
  26. #include "libavcodec/avcodec.h"
  27. #include "avfilter.h"
  28. #include "internal.h"
  29. typedef struct {
  30. struct AVResampleContext *resample;
  31. int out_rate;
  32. double ratio;
  33. AVFilterBufferRef *outsamplesref;
  34. int unconsumed_nb_samples,
  35. max_cached_nb_samples;
  36. int16_t *cached_data[8],
  37. *resampled_data[8];
  38. } AResampleContext;
  39. static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
  40. {
  41. AResampleContext *aresample = ctx->priv;
  42. int ret;
  43. if (args) {
  44. if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0)
  45. return ret;
  46. } else {
  47. aresample->out_rate = -1;
  48. }
  49. return 0;
  50. }
  51. static av_cold void uninit(AVFilterContext *ctx)
  52. {
  53. AResampleContext *aresample = ctx->priv;
  54. if (aresample->outsamplesref) {
  55. int nb_channels =
  56. av_get_channel_layout_nb_channels(
  57. aresample->outsamplesref->audio->channel_layout);
  58. avfilter_unref_buffer(aresample->outsamplesref);
  59. while (nb_channels--) {
  60. av_freep(&(aresample->cached_data[nb_channels]));
  61. av_freep(&(aresample->resampled_data[nb_channels]));
  62. }
  63. }
  64. if (aresample->resample)
  65. av_resample_close(aresample->resample);
  66. }
  67. static int config_output(AVFilterLink *outlink)
  68. {
  69. AVFilterContext *ctx = outlink->src;
  70. AVFilterLink *inlink = ctx->inputs[0];
  71. AResampleContext *aresample = ctx->priv;
  72. if (aresample->out_rate == -1)
  73. aresample->out_rate = outlink->sample_rate;
  74. else
  75. outlink->sample_rate = aresample->out_rate;
  76. //TODO: make the resampling parameters configurable
  77. aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
  78. 16, 10, 0, 0.8);
  79. aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
  80. av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
  81. inlink->sample_rate, outlink->sample_rate);
  82. return 0;
  83. }
  84. static int query_formats(AVFilterContext *ctx)
  85. {
  86. AVFilterFormats *formats = NULL;
  87. avfilter_add_format(&formats, AV_SAMPLE_FMT_S16);
  88. if (!formats)
  89. return AVERROR(ENOMEM);
  90. avfilter_set_common_sample_formats(ctx, formats);
  91. formats = avfilter_all_channel_layouts();
  92. if (!formats)
  93. return AVERROR(ENOMEM);
  94. avfilter_set_common_channel_layouts(ctx, formats);
  95. formats = avfilter_all_packing_formats();
  96. if (!formats)
  97. return AVERROR(ENOMEM);
  98. avfilter_set_common_packing_formats(ctx, formats);
  99. return 0;
  100. }
  101. static void deinterleave(int16_t **outp, int16_t *in,
  102. int nb_channels, int nb_samples)
  103. {
  104. int16_t *out[8];
  105. memcpy(out, outp, nb_channels * sizeof(int16_t*));
  106. switch (nb_channels) {
  107. case 2:
  108. while (nb_samples--) {
  109. *out[0]++ = *in++;
  110. *out[1]++ = *in++;
  111. }
  112. break;
  113. case 3:
  114. while (nb_samples--) {
  115. *out[0]++ = *in++;
  116. *out[1]++ = *in++;
  117. *out[2]++ = *in++;
  118. }
  119. break;
  120. case 4:
  121. while (nb_samples--) {
  122. *out[0]++ = *in++;
  123. *out[1]++ = *in++;
  124. *out[2]++ = *in++;
  125. *out[3]++ = *in++;
  126. }
  127. break;
  128. case 5:
  129. while (nb_samples--) {
  130. *out[0]++ = *in++;
  131. *out[1]++ = *in++;
  132. *out[2]++ = *in++;
  133. *out[3]++ = *in++;
  134. *out[4]++ = *in++;
  135. }
  136. break;
  137. case 6:
  138. while (nb_samples--) {
  139. *out[0]++ = *in++;
  140. *out[1]++ = *in++;
  141. *out[2]++ = *in++;
  142. *out[3]++ = *in++;
  143. *out[4]++ = *in++;
  144. *out[5]++ = *in++;
  145. }
  146. break;
  147. case 8:
  148. while (nb_samples--) {
  149. *out[0]++ = *in++;
  150. *out[1]++ = *in++;
  151. *out[2]++ = *in++;
  152. *out[3]++ = *in++;
  153. *out[4]++ = *in++;
  154. *out[5]++ = *in++;
  155. *out[6]++ = *in++;
  156. *out[7]++ = *in++;
  157. }
  158. break;
  159. }
  160. }
  161. static void interleave(int16_t *out, int16_t **inp,
  162. int nb_channels, int nb_samples)
  163. {
  164. int16_t *in[8];
  165. memcpy(in, inp, nb_channels * sizeof(int16_t*));
  166. switch (nb_channels) {
  167. case 2:
  168. while (nb_samples--) {
  169. *out++ = *in[0]++;
  170. *out++ = *in[1]++;
  171. }
  172. break;
  173. case 3:
  174. while (nb_samples--) {
  175. *out++ = *in[0]++;
  176. *out++ = *in[1]++;
  177. *out++ = *in[2]++;
  178. }
  179. break;
  180. case 4:
  181. while (nb_samples--) {
  182. *out++ = *in[0]++;
  183. *out++ = *in[1]++;
  184. *out++ = *in[2]++;
  185. *out++ = *in[3]++;
  186. }
  187. break;
  188. case 5:
  189. while (nb_samples--) {
  190. *out++ = *in[0]++;
  191. *out++ = *in[1]++;
  192. *out++ = *in[2]++;
  193. *out++ = *in[3]++;
  194. *out++ = *in[4]++;
  195. }
  196. break;
  197. case 6:
  198. while (nb_samples--) {
  199. *out++ = *in[0]++;
  200. *out++ = *in[1]++;
  201. *out++ = *in[2]++;
  202. *out++ = *in[3]++;
  203. *out++ = *in[4]++;
  204. *out++ = *in[5]++;
  205. }
  206. break;
  207. case 8:
  208. while (nb_samples--) {
  209. *out++ = *in[0]++;
  210. *out++ = *in[1]++;
  211. *out++ = *in[2]++;
  212. *out++ = *in[3]++;
  213. *out++ = *in[4]++;
  214. *out++ = *in[5]++;
  215. *out++ = *in[6]++;
  216. *out++ = *in[7]++;
  217. }
  218. break;
  219. }
  220. }
  221. static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
  222. {
  223. AResampleContext *aresample = inlink->dst->priv;
  224. AVFilterLink * const outlink = inlink->dst->outputs[0];
  225. int i,
  226. in_nb_samples = insamplesref->audio->nb_samples,
  227. cached_nb_samples = in_nb_samples + aresample->unconsumed_nb_samples,
  228. requested_out_nb_samples = aresample->ratio * cached_nb_samples,
  229. nb_channels =
  230. av_get_channel_layout_nb_channels(inlink->channel_layout);
  231. if (cached_nb_samples > aresample->max_cached_nb_samples) {
  232. for (i = 0; i < nb_channels; i++) {
  233. aresample->cached_data[i] =
  234. av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
  235. aresample->resampled_data[i] =
  236. av_realloc(aresample->resampled_data[i],
  237. FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
  238. if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
  239. return;
  240. }
  241. aresample->max_cached_nb_samples = cached_nb_samples;
  242. if (aresample->outsamplesref)
  243. avfilter_unref_buffer(aresample->outsamplesref);
  244. aresample->outsamplesref = avfilter_get_audio_buffer(outlink,
  245. AV_PERM_WRITE | AV_PERM_REUSE2,
  246. inlink->format,
  247. requested_out_nb_samples,
  248. insamplesref->audio->channel_layout,
  249. insamplesref->audio->planar);
  250. avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
  251. aresample->outsamplesref->pts =
  252. insamplesref->pts / inlink->sample_rate * outlink->sample_rate;
  253. aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
  254. outlink->out_buf = aresample->outsamplesref;
  255. }
  256. /* av_resample() works with planar audio buffers */
  257. if (!inlink->planar && nb_channels > 1) {
  258. int16_t *out[8];
  259. for (i = 0; i < nb_channels; i++)
  260. out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
  261. deinterleave(out, (int16_t *)insamplesref->data[0],
  262. nb_channels, in_nb_samples);
  263. } else {
  264. for (i = 0; i < nb_channels; i++)
  265. memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
  266. insamplesref->data[i],
  267. in_nb_samples * sizeof(int16_t));
  268. }
  269. for (i = 0; i < nb_channels; i++) {
  270. int consumed_nb_samples;
  271. const int is_last = i+1 == nb_channels;
  272. aresample->outsamplesref->audio->nb_samples =
  273. av_resample(aresample->resample,
  274. aresample->resampled_data[i], aresample->cached_data[i],
  275. &consumed_nb_samples,
  276. cached_nb_samples,
  277. requested_out_nb_samples, is_last);
  278. /* move unconsumed data back to the beginning of the cache */
  279. aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
  280. memmove(aresample->cached_data[i],
  281. aresample->cached_data[i] + consumed_nb_samples,
  282. aresample->unconsumed_nb_samples * sizeof(int16_t));
  283. }
  284. /* copy resampled data to the output samplesref */
  285. if (!inlink->planar && nb_channels > 1) {
  286. interleave((int16_t *)aresample->outsamplesref->data[0],
  287. aresample->resampled_data,
  288. nb_channels, aresample->outsamplesref->audio->nb_samples);
  289. } else {
  290. for (i = 0; i < nb_channels; i++)
  291. memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
  292. aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
  293. }
  294. avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
  295. avfilter_unref_buffer(insamplesref);
  296. }
  297. AVFilter avfilter_af_aresample = {
  298. .name = "aresample",
  299. .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
  300. .init = init,
  301. .uninit = uninit,
  302. .query_formats = query_formats,
  303. .priv_size = sizeof(AResampleContext),
  304. .inputs = (AVFilterPad[]) {{ .name = "default",
  305. .type = AVMEDIA_TYPE_AUDIO,
  306. .filter_samples = filter_samples,
  307. .min_perms = AV_PERM_READ, },
  308. { .name = NULL}},
  309. .outputs = (AVFilterPad[]) {{ .name = "default",
  310. .config_props = config_output,
  311. .type = AVMEDIA_TYPE_AUDIO, },
  312. { .name = NULL}},
  313. };