af_amerge.c 12 KB

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  1. /*
  2. * Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
  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
  14. * GNU 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. /**
  21. * @file
  22. * Audio merging filter
  23. */
  24. #include "libavutil/audioconvert.h"
  25. #include "libavutil/bprint.h"
  26. #include "libavutil/opt.h"
  27. #include "libswresample/swresample.h" // only for SWR_CH_MAX
  28. #include "avfilter.h"
  29. #include "audio.h"
  30. #include "bufferqueue.h"
  31. #include "internal.h"
  32. typedef struct {
  33. const AVClass *class;
  34. int nb_inputs;
  35. int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
  36. int bps;
  37. struct amerge_input {
  38. struct FFBufQueue queue;
  39. int nb_ch; /**< number of channels for the input */
  40. int nb_samples;
  41. int pos;
  42. } *in;
  43. } AMergeContext;
  44. #define OFFSET(x) offsetof(AMergeContext, x)
  45. static const AVOption amerge_options[] = {
  46. { "inputs", "specify the number of inputs", OFFSET(nb_inputs),
  47. AV_OPT_TYPE_INT, { .dbl = 2 }, 2, SWR_CH_MAX },
  48. {0}
  49. };
  50. AVFILTER_DEFINE_CLASS(amerge);
  51. static av_cold void uninit(AVFilterContext *ctx)
  52. {
  53. AMergeContext *am = ctx->priv;
  54. int i;
  55. for (i = 0; i < am->nb_inputs; i++)
  56. ff_bufqueue_discard_all(&am->in[i].queue);
  57. av_freep(&am->in);
  58. }
  59. static int query_formats(AVFilterContext *ctx)
  60. {
  61. AMergeContext *am = ctx->priv;
  62. int64_t inlayout[SWR_CH_MAX], outlayout = 0;
  63. AVFilterFormats *formats;
  64. AVFilterChannelLayouts *layouts;
  65. int i, overlap = 0, nb_ch = 0;
  66. for (i = 0; i < am->nb_inputs; i++) {
  67. if (!ctx->inputs[i]->in_channel_layouts ||
  68. !ctx->inputs[i]->in_channel_layouts->nb_channel_layouts) {
  69. av_log(ctx, AV_LOG_ERROR,
  70. "No channel layout for input %d\n", i + 1);
  71. return AVERROR(EINVAL);
  72. }
  73. inlayout[i] = ctx->inputs[i]->in_channel_layouts->channel_layouts[0];
  74. if (ctx->inputs[i]->in_channel_layouts->nb_channel_layouts > 1) {
  75. char buf[256];
  76. av_get_channel_layout_string(buf, sizeof(buf), 0, inlayout[i]);
  77. av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
  78. }
  79. am->in[i].nb_ch = av_get_channel_layout_nb_channels(inlayout[i]);
  80. if (outlayout & inlayout[i])
  81. overlap++;
  82. outlayout |= inlayout[i];
  83. nb_ch += am->in[i].nb_ch;
  84. }
  85. if (nb_ch > SWR_CH_MAX) {
  86. av_log(ctx, AV_LOG_ERROR, "Too many channels (max %d)\n", SWR_CH_MAX);
  87. return AVERROR(EINVAL);
  88. }
  89. if (overlap) {
  90. av_log(ctx, AV_LOG_WARNING,
  91. "Inputs overlap: output layout will be meaningless\n");
  92. for (i = 0; i < nb_ch; i++)
  93. am->route[i] = i;
  94. outlayout = av_get_default_channel_layout(nb_ch);
  95. if (!outlayout)
  96. outlayout = ((int64_t)1 << nb_ch) - 1;
  97. } else {
  98. int *route[SWR_CH_MAX];
  99. int c, out_ch_number = 0;
  100. route[0] = am->route;
  101. for (i = 1; i < am->nb_inputs; i++)
  102. route[i] = route[i - 1] + am->in[i - 1].nb_ch;
  103. for (c = 0; c < 64; c++)
  104. for (i = 0; i < am->nb_inputs; i++)
  105. if ((inlayout[i] >> c) & 1)
  106. *(route[i]++) = out_ch_number++;
  107. }
  108. formats = ff_make_format_list(ff_packed_sample_fmts_array);
  109. ff_set_common_formats(ctx, formats);
  110. for (i = 0; i < am->nb_inputs; i++) {
  111. layouts = NULL;
  112. ff_add_channel_layout(&layouts, inlayout[i]);
  113. ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);
  114. }
  115. layouts = NULL;
  116. ff_add_channel_layout(&layouts, outlayout);
  117. ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
  118. ff_set_common_samplerates(ctx, ff_all_samplerates());
  119. return 0;
  120. }
  121. static int config_output(AVFilterLink *outlink)
  122. {
  123. AVFilterContext *ctx = outlink->src;
  124. AMergeContext *am = ctx->priv;
  125. AVBPrint bp;
  126. int i;
  127. for (i = 1; i < am->nb_inputs; i++) {
  128. if (ctx->inputs[i]->sample_rate != ctx->inputs[0]->sample_rate) {
  129. av_log(ctx, AV_LOG_ERROR,
  130. "Inputs must have the same sample rate "
  131. "%d for in%d vs %d\n",
  132. ctx->inputs[i]->sample_rate, i, ctx->inputs[0]->sample_rate);
  133. return AVERROR(EINVAL);
  134. }
  135. }
  136. am->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
  137. outlink->sample_rate = ctx->inputs[0]->sample_rate;
  138. outlink->time_base = ctx->inputs[0]->time_base;
  139. av_bprint_init(&bp, 0, 1);
  140. for (i = 0; i < am->nb_inputs; i++) {
  141. av_bprintf(&bp, "%sin%d:", i ? " + " : "", i);
  142. av_bprint_channel_layout(&bp, -1, ctx->inputs[i]->channel_layout);
  143. }
  144. av_bprintf(&bp, " -> out:");
  145. av_bprint_channel_layout(&bp, -1, ctx->outputs[0]->channel_layout);
  146. av_log(ctx, AV_LOG_INFO, "%s\n", bp.str);
  147. return 0;
  148. }
  149. static int request_frame(AVFilterLink *outlink)
  150. {
  151. AVFilterContext *ctx = outlink->src;
  152. AMergeContext *am = ctx->priv;
  153. int i, ret;
  154. for (i = 0; i < am->nb_inputs; i++)
  155. if (!am->in[i].nb_samples)
  156. if ((ret = ff_request_frame(ctx->inputs[i])) < 0)
  157. return ret;
  158. return 0;
  159. }
  160. /**
  161. * Copy samples from several input streams to one output stream.
  162. * @param nb_inputs number of inputs
  163. * @param in inputs; used only for the nb_ch field;
  164. * @param route routing values;
  165. * input channel i goes to output channel route[i];
  166. * i < in[0].nb_ch are the channels from the first output;
  167. * i >= in[0].nb_ch are the channels from the second output
  168. * @param ins pointer to the samples of each inputs, in packed format;
  169. * will be left at the end of the copied samples
  170. * @param outs pointer to the samples of the output, in packet format;
  171. * must point to a buffer big enough;
  172. * will be left at the end of the copied samples
  173. * @param ns number of samples to copy
  174. * @param bps bytes per sample
  175. */
  176. static inline void copy_samples(int nb_inputs, struct amerge_input in[],
  177. int *route, uint8_t *ins[],
  178. uint8_t **outs, int ns, int bps)
  179. {
  180. int *route_cur;
  181. int i, c, nb_ch = 0;
  182. for (i = 0; i < nb_inputs; i++)
  183. nb_ch += in[i].nb_ch;
  184. while (ns--) {
  185. route_cur = route;
  186. for (i = 0; i < nb_inputs; i++) {
  187. for (c = 0; c < in[i].nb_ch; c++) {
  188. memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
  189. ins[i] += bps;
  190. }
  191. }
  192. *outs += nb_ch * bps;
  193. }
  194. }
  195. static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
  196. {
  197. AVFilterContext *ctx = inlink->dst;
  198. AMergeContext *am = ctx->priv;
  199. AVFilterLink *const outlink = ctx->outputs[0];
  200. int input_number;
  201. int nb_samples, ns, i;
  202. AVFilterBufferRef *outbuf, *inbuf[SWR_CH_MAX];
  203. uint8_t *ins[SWR_CH_MAX], *outs;
  204. for (input_number = 0; input_number < am->nb_inputs; input_number++)
  205. if (inlink == ctx->inputs[input_number])
  206. break;
  207. av_assert1(input_number < am->nb_inputs);
  208. ff_bufqueue_add(ctx, &am->in[input_number].queue, insamples);
  209. am->in[input_number].nb_samples += insamples->audio->nb_samples;
  210. nb_samples = am->in[0].nb_samples;
  211. for (i = 1; i < am->nb_inputs; i++)
  212. nb_samples = FFMIN(nb_samples, am->in[i].nb_samples);
  213. if (!nb_samples)
  214. return 0;
  215. outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
  216. outs = outbuf->data[0];
  217. for (i = 0; i < am->nb_inputs; i++) {
  218. inbuf[i] = ff_bufqueue_peek(&am->in[i].queue, 0);
  219. ins[i] = inbuf[i]->data[0] +
  220. am->in[i].pos * am->in[i].nb_ch * am->bps;
  221. }
  222. outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
  223. inbuf[0]->pts +
  224. av_rescale_q(am->in[0].pos,
  225. (AVRational){ 1, ctx->inputs[0]->sample_rate },
  226. ctx->outputs[0]->time_base);
  227. avfilter_copy_buffer_ref_props(outbuf, inbuf[0]);
  228. outbuf->audio->nb_samples = nb_samples;
  229. outbuf->audio->channel_layout = outlink->channel_layout;
  230. while (nb_samples) {
  231. ns = nb_samples;
  232. for (i = 0; i < am->nb_inputs; i++)
  233. ns = FFMIN(ns, inbuf[i]->audio->nb_samples - am->in[i].pos);
  234. /* Unroll the most common sample formats: speed +~350% for the loop,
  235. +~13% overall (including two common decoders) */
  236. switch (am->bps) {
  237. case 1:
  238. copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, 1);
  239. break;
  240. case 2:
  241. copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, 2);
  242. break;
  243. case 4:
  244. copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, 4);
  245. break;
  246. default:
  247. copy_samples(am->nb_inputs, am->in, am->route, ins, &outs, ns, am->bps);
  248. break;
  249. }
  250. nb_samples -= ns;
  251. for (i = 0; i < am->nb_inputs; i++) {
  252. am->in[i].nb_samples -= ns;
  253. am->in[i].pos += ns;
  254. if (am->in[i].pos == inbuf[i]->audio->nb_samples) {
  255. am->in[i].pos = 0;
  256. avfilter_unref_buffer(inbuf[i]);
  257. ff_bufqueue_get(&am->in[i].queue);
  258. inbuf[i] = ff_bufqueue_peek(&am->in[i].queue, 0);
  259. ins[i] = inbuf[i] ? inbuf[i]->data[0] : NULL;
  260. }
  261. }
  262. }
  263. return ff_filter_samples(ctx->outputs[0], outbuf);
  264. }
  265. static av_cold int init(AVFilterContext *ctx, const char *args)
  266. {
  267. AMergeContext *am = ctx->priv;
  268. int ret, i;
  269. char name[16];
  270. am->class = &amerge_class;
  271. av_opt_set_defaults(am);
  272. ret = av_set_options_string(am, args, "=", ":");
  273. if (ret < 0) {
  274. av_log(ctx, AV_LOG_ERROR, "Error parsing options: '%s'\n", args);
  275. return ret;
  276. }
  277. am->in = av_calloc(am->nb_inputs, sizeof(*am->in));
  278. if (!am->in)
  279. return AVERROR(ENOMEM);
  280. for (i = 0; i < am->nb_inputs; i++) {
  281. AVFilterPad pad = {
  282. .name = name,
  283. .type = AVMEDIA_TYPE_AUDIO,
  284. .filter_samples = filter_samples,
  285. .min_perms = AV_PERM_READ | AV_PERM_PRESERVE,
  286. };
  287. snprintf(name, sizeof(name), "in%d", i);
  288. ff_insert_inpad(ctx, i, &pad);
  289. }
  290. return 0;
  291. }
  292. AVFilter avfilter_af_amerge = {
  293. .name = "amerge",
  294. .description = NULL_IF_CONFIG_SMALL("Merge two audio streams into "
  295. "a single multi-channel stream."),
  296. .priv_size = sizeof(AMergeContext),
  297. .init = init,
  298. .uninit = uninit,
  299. .query_formats = query_formats,
  300. .inputs = (const AVFilterPad[]) { { .name = NULL } },
  301. .outputs = (const AVFilterPad[]) {
  302. { .name = "default",
  303. .type = AVMEDIA_TYPE_AUDIO,
  304. .config_props = config_output,
  305. .request_frame = request_frame, },
  306. { .name = NULL }
  307. },
  308. };