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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/avstring.h"
  26. #include "libavutil/channel_layout.h"
  27. #include "libavutil/opt.h"
  28. #include "libavutil/samplefmt.h"
  29. #include "libavutil/avassert.h"
  30. #include "libswresample/swresample.h"
  31. #include "avfilter.h"
  32. #include "audio.h"
  33. #include "internal.h"
  34. typedef struct {
  35. const AVClass *class;
  36. int sample_rate_arg;
  37. double ratio;
  38. struct SwrContext *swr;
  39. int64_t next_pts;
  40. int req_fullfilled;
  41. } AResampleContext;
  42. static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
  43. {
  44. AResampleContext *aresample = ctx->priv;
  45. int ret = 0;
  46. aresample->next_pts = AV_NOPTS_VALUE;
  47. aresample->swr = swr_alloc();
  48. if (!aresample->swr) {
  49. ret = AVERROR(ENOMEM);
  50. goto end;
  51. }
  52. if (opts) {
  53. AVDictionaryEntry *e = NULL;
  54. while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
  55. if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
  56. goto end;
  57. }
  58. av_dict_free(opts);
  59. }
  60. if (aresample->sample_rate_arg > 0)
  61. av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
  62. end:
  63. return ret;
  64. }
  65. static av_cold void uninit(AVFilterContext *ctx)
  66. {
  67. AResampleContext *aresample = ctx->priv;
  68. swr_free(&aresample->swr);
  69. }
  70. static int query_formats(AVFilterContext *ctx)
  71. {
  72. AResampleContext *aresample = ctx->priv;
  73. int out_rate = av_get_int(aresample->swr, "osr", NULL);
  74. uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
  75. enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
  76. AVFilterLink *inlink = ctx->inputs[0];
  77. AVFilterLink *outlink = ctx->outputs[0];
  78. AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  79. AVFilterFormats *out_formats;
  80. AVFilterFormats *in_samplerates = ff_all_samplerates();
  81. AVFilterFormats *out_samplerates;
  82. AVFilterChannelLayouts *in_layouts = ff_all_channel_counts();
  83. AVFilterChannelLayouts *out_layouts;
  84. ff_formats_ref (in_formats, &inlink->out_formats);
  85. ff_formats_ref (in_samplerates, &inlink->out_samplerates);
  86. ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
  87. if(out_rate > 0) {
  88. out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
  89. } else {
  90. out_samplerates = ff_all_samplerates();
  91. }
  92. ff_formats_ref(out_samplerates, &outlink->in_samplerates);
  93. if(out_format != AV_SAMPLE_FMT_NONE) {
  94. out_formats = ff_make_format_list((int[]){ out_format, -1 });
  95. } else
  96. out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  97. ff_formats_ref(out_formats, &outlink->in_formats);
  98. if(out_layout) {
  99. out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
  100. } else
  101. out_layouts = ff_all_channel_counts();
  102. ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
  103. return 0;
  104. }
  105. static int config_output(AVFilterLink *outlink)
  106. {
  107. int ret;
  108. AVFilterContext *ctx = outlink->src;
  109. AVFilterLink *inlink = ctx->inputs[0];
  110. AResampleContext *aresample = ctx->priv;
  111. int out_rate;
  112. uint64_t out_layout;
  113. enum AVSampleFormat out_format;
  114. char inchl_buf[128], outchl_buf[128];
  115. aresample->swr = swr_alloc_set_opts(aresample->swr,
  116. outlink->channel_layout, outlink->format, outlink->sample_rate,
  117. inlink->channel_layout, inlink->format, inlink->sample_rate,
  118. 0, ctx);
  119. if (!aresample->swr)
  120. return AVERROR(ENOMEM);
  121. if (!inlink->channel_layout)
  122. av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
  123. if (!outlink->channel_layout)
  124. av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
  125. ret = swr_init(aresample->swr);
  126. if (ret < 0)
  127. return ret;
  128. out_rate = av_get_int(aresample->swr, "osr", NULL);
  129. out_layout = av_get_int(aresample->swr, "ocl", NULL);
  130. out_format = av_get_int(aresample->swr, "osf", NULL);
  131. outlink->time_base = (AVRational) {1, out_rate};
  132. av_assert0(outlink->sample_rate == out_rate);
  133. av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
  134. av_assert0(outlink->format == out_format);
  135. aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
  136. av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
  137. av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
  138. av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
  139. inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
  140. outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
  141. return 0;
  142. }
  143. static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
  144. {
  145. AResampleContext *aresample = inlink->dst->priv;
  146. const int n_in = insamplesref->nb_samples;
  147. int n_out = n_in * aresample->ratio * 2 + 256;
  148. AVFilterLink *const outlink = inlink->dst->outputs[0];
  149. AVFrame *outsamplesref = ff_get_audio_buffer(outlink, n_out);
  150. int ret;
  151. if(!outsamplesref)
  152. return AVERROR(ENOMEM);
  153. av_frame_copy_props(outsamplesref, insamplesref);
  154. outsamplesref->format = outlink->format;
  155. av_frame_set_channels(outsamplesref, outlink->channels);
  156. outsamplesref->channel_layout = outlink->channel_layout;
  157. outsamplesref->sample_rate = outlink->sample_rate;
  158. if(insamplesref->pts != AV_NOPTS_VALUE) {
  159. int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
  160. int64_t outpts= swr_next_pts(aresample->swr, inpts);
  161. aresample->next_pts =
  162. outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
  163. } else {
  164. outsamplesref->pts = AV_NOPTS_VALUE;
  165. }
  166. n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
  167. (void *)insamplesref->extended_data, n_in);
  168. if (n_out <= 0) {
  169. av_frame_free(&outsamplesref);
  170. av_frame_free(&insamplesref);
  171. return 0;
  172. }
  173. outsamplesref->nb_samples = n_out;
  174. ret = ff_filter_frame(outlink, outsamplesref);
  175. aresample->req_fullfilled= 1;
  176. av_frame_free(&insamplesref);
  177. return ret;
  178. }
  179. static int request_frame(AVFilterLink *outlink)
  180. {
  181. AVFilterContext *ctx = outlink->src;
  182. AResampleContext *aresample = ctx->priv;
  183. AVFilterLink *const inlink = outlink->src->inputs[0];
  184. int ret;
  185. aresample->req_fullfilled = 0;
  186. do{
  187. ret = ff_request_frame(ctx->inputs[0]);
  188. }while(!aresample->req_fullfilled && ret>=0);
  189. if (ret == AVERROR_EOF) {
  190. AVFrame *outsamplesref;
  191. int n_out = 4096;
  192. outsamplesref = ff_get_audio_buffer(outlink, n_out);
  193. if (!outsamplesref)
  194. return AVERROR(ENOMEM);
  195. n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
  196. if (n_out <= 0) {
  197. av_frame_free(&outsamplesref);
  198. return (n_out == 0) ? AVERROR_EOF : n_out;
  199. }
  200. outsamplesref->sample_rate = outlink->sample_rate;
  201. outsamplesref->nb_samples = n_out;
  202. #if 0
  203. outsamplesref->pts = aresample->next_pts;
  204. if(aresample->next_pts != AV_NOPTS_VALUE)
  205. aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
  206. #else
  207. outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN);
  208. outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
  209. #endif
  210. return ff_filter_frame(outlink, outsamplesref);
  211. }
  212. return ret;
  213. }
  214. static const AVClass *resample_child_class_next(const AVClass *prev)
  215. {
  216. return prev ? NULL : swr_get_class();
  217. }
  218. static void *resample_child_next(void *obj, void *prev)
  219. {
  220. AResampleContext *s = obj;
  221. return prev ? NULL : s->swr;
  222. }
  223. #define OFFSET(x) offsetof(AResampleContext, x)
  224. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  225. static const AVOption options[] = {
  226. {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
  227. {NULL}
  228. };
  229. static const AVClass aresample_class = {
  230. .class_name = "aresample",
  231. .item_name = av_default_item_name,
  232. .option = options,
  233. .version = LIBAVUTIL_VERSION_INT,
  234. .child_class_next = resample_child_class_next,
  235. .child_next = resample_child_next,
  236. };
  237. static const AVFilterPad aresample_inputs[] = {
  238. {
  239. .name = "default",
  240. .type = AVMEDIA_TYPE_AUDIO,
  241. .filter_frame = filter_frame,
  242. },
  243. { NULL }
  244. };
  245. static const AVFilterPad aresample_outputs[] = {
  246. {
  247. .name = "default",
  248. .config_props = config_output,
  249. .request_frame = request_frame,
  250. .type = AVMEDIA_TYPE_AUDIO,
  251. },
  252. { NULL }
  253. };
  254. AVFilter avfilter_af_aresample = {
  255. .name = "aresample",
  256. .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
  257. .init_dict = init_dict,
  258. .uninit = uninit,
  259. .query_formats = query_formats,
  260. .priv_size = sizeof(AResampleContext),
  261. .priv_class = &aresample_class,
  262. .inputs = aresample_inputs,
  263. .outputs = aresample_outputs,
  264. };