af_resample.c 11 KB

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  1. /*
  2. * This file is part of FFmpeg.
  3. *
  4. * FFmpeg is free software; you can redistribute it and/or
  5. * modify it under the terms of the GNU Lesser General Public
  6. * License as published by the Free Software Foundation; either
  7. * version 2.1 of the License, or (at your option) any later version.
  8. *
  9. * FFmpeg is distributed in the hope that it will be useful,
  10. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  11. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  12. * Lesser General Public License for more details.
  13. *
  14. * You should have received a copy of the GNU Lesser General Public
  15. * License along with FFmpeg; if not, write to the Free Software
  16. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  17. */
  18. /**
  19. * @file
  20. * sample format and channel layout conversion audio filter
  21. */
  22. #include "libavutil/avassert.h"
  23. #include "libavutil/avstring.h"
  24. #include "libavutil/common.h"
  25. #include "libavutil/dict.h"
  26. #include "libavutil/mathematics.h"
  27. #include "libavutil/opt.h"
  28. #include "libavresample/avresample.h"
  29. #include "audio.h"
  30. #include "avfilter.h"
  31. #include "formats.h"
  32. #include "internal.h"
  33. typedef struct ResampleContext {
  34. const AVClass *class;
  35. AVAudioResampleContext *avr;
  36. AVDictionary *options;
  37. int64_t next_pts;
  38. int64_t next_in_pts;
  39. /* set by filter_frame() to signal an output frame to request_frame() */
  40. int got_output;
  41. } ResampleContext;
  42. static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
  43. {
  44. ResampleContext *s = ctx->priv;
  45. const AVClass *avr_class = avresample_get_class();
  46. AVDictionaryEntry *e = NULL;
  47. while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
  48. if (av_opt_find(&avr_class, e->key, NULL, 0,
  49. AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))
  50. av_dict_set(&s->options, e->key, e->value, 0);
  51. }
  52. e = NULL;
  53. while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
  54. av_dict_set(opts, e->key, NULL, 0);
  55. /* do not allow the user to override basic format options */
  56. av_dict_set(&s->options, "in_channel_layout", NULL, 0);
  57. av_dict_set(&s->options, "out_channel_layout", NULL, 0);
  58. av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
  59. av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
  60. av_dict_set(&s->options, "in_sample_rate", NULL, 0);
  61. av_dict_set(&s->options, "out_sample_rate", NULL, 0);
  62. return 0;
  63. }
  64. static av_cold void uninit(AVFilterContext *ctx)
  65. {
  66. ResampleContext *s = ctx->priv;
  67. if (s->avr) {
  68. avresample_close(s->avr);
  69. avresample_free(&s->avr);
  70. }
  71. av_dict_free(&s->options);
  72. }
  73. static int query_formats(AVFilterContext *ctx)
  74. {
  75. AVFilterLink *inlink = ctx->inputs[0];
  76. AVFilterLink *outlink = ctx->outputs[0];
  77. AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  78. AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  79. AVFilterFormats *in_samplerates = ff_all_samplerates();
  80. AVFilterFormats *out_samplerates = ff_all_samplerates();
  81. AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
  82. AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
  83. ff_formats_ref(in_formats, &inlink->out_formats);
  84. ff_formats_ref(out_formats, &outlink->in_formats);
  85. ff_formats_ref(in_samplerates, &inlink->out_samplerates);
  86. ff_formats_ref(out_samplerates, &outlink->in_samplerates);
  87. ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
  88. ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
  89. return 0;
  90. }
  91. static int config_output(AVFilterLink *outlink)
  92. {
  93. AVFilterContext *ctx = outlink->src;
  94. AVFilterLink *inlink = ctx->inputs[0];
  95. ResampleContext *s = ctx->priv;
  96. char buf1[64], buf2[64];
  97. int ret;
  98. if (s->avr) {
  99. avresample_close(s->avr);
  100. avresample_free(&s->avr);
  101. }
  102. if (inlink->channel_layout == outlink->channel_layout &&
  103. inlink->sample_rate == outlink->sample_rate &&
  104. (inlink->format == outlink->format ||
  105. (av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
  106. av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
  107. av_get_planar_sample_fmt(inlink->format) ==
  108. av_get_planar_sample_fmt(outlink->format))))
  109. return 0;
  110. if (!(s->avr = avresample_alloc_context()))
  111. return AVERROR(ENOMEM);
  112. if (s->options) {
  113. int ret;
  114. AVDictionaryEntry *e = NULL;
  115. while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
  116. av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
  117. ret = av_opt_set_dict(s->avr, &s->options);
  118. if (ret < 0)
  119. return ret;
  120. }
  121. av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
  122. av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
  123. av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
  124. av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
  125. av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
  126. av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
  127. if ((ret = avresample_open(s->avr)) < 0)
  128. return ret;
  129. outlink->time_base = (AVRational){ 1, outlink->sample_rate };
  130. s->next_pts = AV_NOPTS_VALUE;
  131. s->next_in_pts = AV_NOPTS_VALUE;
  132. av_get_channel_layout_string(buf1, sizeof(buf1),
  133. -1, inlink ->channel_layout);
  134. av_get_channel_layout_string(buf2, sizeof(buf2),
  135. -1, outlink->channel_layout);
  136. av_log(ctx, AV_LOG_VERBOSE,
  137. "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
  138. av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
  139. av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
  140. return 0;
  141. }
  142. static int request_frame(AVFilterLink *outlink)
  143. {
  144. AVFilterContext *ctx = outlink->src;
  145. ResampleContext *s = ctx->priv;
  146. int ret = 0;
  147. s->got_output = 0;
  148. while (ret >= 0 && !s->got_output)
  149. ret = ff_request_frame(ctx->inputs[0]);
  150. /* flush the lavr delay buffer */
  151. if (ret == AVERROR_EOF && s->avr) {
  152. AVFrame *frame;
  153. int nb_samples = avresample_get_out_samples(s->avr, 0);
  154. if (!nb_samples)
  155. return ret;
  156. frame = ff_get_audio_buffer(outlink, nb_samples);
  157. if (!frame)
  158. return AVERROR(ENOMEM);
  159. ret = avresample_convert(s->avr, frame->extended_data,
  160. frame->linesize[0], nb_samples,
  161. NULL, 0, 0);
  162. if (ret <= 0) {
  163. av_frame_free(&frame);
  164. return (ret == 0) ? AVERROR_EOF : ret;
  165. }
  166. frame->pts = s->next_pts;
  167. return ff_filter_frame(outlink, frame);
  168. }
  169. return ret;
  170. }
  171. static int filter_frame(AVFilterLink *inlink, AVFrame *in)
  172. {
  173. AVFilterContext *ctx = inlink->dst;
  174. ResampleContext *s = ctx->priv;
  175. AVFilterLink *outlink = ctx->outputs[0];
  176. int ret;
  177. if (s->avr) {
  178. AVFrame *out;
  179. int delay, nb_samples;
  180. /* maximum possible samples lavr can output */
  181. delay = avresample_get_delay(s->avr);
  182. nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
  183. out = ff_get_audio_buffer(outlink, nb_samples);
  184. if (!out) {
  185. ret = AVERROR(ENOMEM);
  186. goto fail;
  187. }
  188. ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
  189. nb_samples, in->extended_data, in->linesize[0],
  190. in->nb_samples);
  191. if (ret <= 0) {
  192. av_frame_free(&out);
  193. if (ret < 0)
  194. goto fail;
  195. }
  196. av_assert0(!avresample_available(s->avr));
  197. if (s->next_pts == AV_NOPTS_VALUE) {
  198. if (in->pts == AV_NOPTS_VALUE) {
  199. av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
  200. "assuming 0.\n");
  201. s->next_pts = 0;
  202. } else
  203. s->next_pts = av_rescale_q(in->pts, inlink->time_base,
  204. outlink->time_base);
  205. }
  206. if (ret > 0) {
  207. out->nb_samples = ret;
  208. ret = av_frame_copy_props(out, in);
  209. if (ret < 0) {
  210. av_frame_free(&out);
  211. goto fail;
  212. }
  213. out->sample_rate = outlink->sample_rate;
  214. /* Only convert in->pts if there is a discontinuous jump.
  215. This ensures that out->pts tracks the number of samples actually
  216. output by the resampler in the absence of such a jump.
  217. Otherwise, the rounding in av_rescale_q() and av_rescale()
  218. causes off-by-1 errors. */
  219. if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
  220. out->pts = av_rescale_q(in->pts, inlink->time_base,
  221. outlink->time_base) -
  222. av_rescale(delay, outlink->sample_rate,
  223. inlink->sample_rate);
  224. } else
  225. out->pts = s->next_pts;
  226. s->next_pts = out->pts + out->nb_samples;
  227. s->next_in_pts = in->pts + in->nb_samples;
  228. ret = ff_filter_frame(outlink, out);
  229. s->got_output = 1;
  230. }
  231. fail:
  232. av_frame_free(&in);
  233. } else {
  234. in->format = outlink->format;
  235. ret = ff_filter_frame(outlink, in);
  236. s->got_output = 1;
  237. }
  238. return ret;
  239. }
  240. static const AVClass *resample_child_class_next(const AVClass *prev)
  241. {
  242. return prev ? NULL : avresample_get_class();
  243. }
  244. static void *resample_child_next(void *obj, void *prev)
  245. {
  246. ResampleContext *s = obj;
  247. return prev ? NULL : s->avr;
  248. }
  249. static const AVClass resample_class = {
  250. .class_name = "resample",
  251. .item_name = av_default_item_name,
  252. .version = LIBAVUTIL_VERSION_INT,
  253. .child_class_next = resample_child_class_next,
  254. .child_next = resample_child_next,
  255. };
  256. static const AVFilterPad avfilter_af_resample_inputs[] = {
  257. {
  258. .name = "default",
  259. .type = AVMEDIA_TYPE_AUDIO,
  260. .filter_frame = filter_frame,
  261. },
  262. { NULL }
  263. };
  264. static const AVFilterPad avfilter_af_resample_outputs[] = {
  265. {
  266. .name = "default",
  267. .type = AVMEDIA_TYPE_AUDIO,
  268. .config_props = config_output,
  269. .request_frame = request_frame
  270. },
  271. { NULL }
  272. };
  273. AVFilter ff_af_resample = {
  274. .name = "resample",
  275. .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
  276. .priv_size = sizeof(ResampleContext),
  277. .priv_class = &resample_class,
  278. .init_dict = init,
  279. .uninit = uninit,
  280. .query_formats = query_formats,
  281. .inputs = avfilter_af_resample_inputs,
  282. .outputs = avfilter_af_resample_outputs,
  283. };