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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. /**
  20.  * @file
  21.  * sample format and channel layout conversion audio filter
  22.  */
  23.  
  24. #include "libavutil/avassert.h"
  25. #include "libavutil/avstring.h"
  26. #include "libavutil/common.h"
  27. #include "libavutil/dict.h"
  28. #include "libavutil/mathematics.h"
  29. #include "libavutil/opt.h"
  30.  
  31. #include "libavresample/avresample.h"
  32.  
  33. #include "audio.h"
  34. #include "avfilter.h"
  35. #include "formats.h"
  36. #include "internal.h"
  37.  
  38. typedef struct ResampleContext {
  39.     const AVClass *class;
  40.     AVAudioResampleContext *avr;
  41.     AVDictionary *options;
  42.  
  43.     int64_t next_pts;
  44.  
  45.     /* set by filter_frame() to signal an output frame to request_frame() */
  46.     int got_output;
  47. } ResampleContext;
  48.  
  49. static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
  50. {
  51.     ResampleContext *s = ctx->priv;
  52.     const AVClass *avr_class = avresample_get_class();
  53.     AVDictionaryEntry *e = NULL;
  54.  
  55.     while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
  56.         if (av_opt_find(&avr_class, e->key, NULL, 0,
  57.                         AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))
  58.             av_dict_set(&s->options, e->key, e->value, 0);
  59.     }
  60.  
  61.     e = NULL;
  62.     while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
  63.         av_dict_set(opts, e->key, NULL, 0);
  64.  
  65.     /* do not allow the user to override basic format options */
  66.     av_dict_set(&s->options,  "in_channel_layout", NULL, 0);
  67.     av_dict_set(&s->options, "out_channel_layout", NULL, 0);
  68.     av_dict_set(&s->options,  "in_sample_fmt",     NULL, 0);
  69.     av_dict_set(&s->options, "out_sample_fmt",     NULL, 0);
  70.     av_dict_set(&s->options,  "in_sample_rate",    NULL, 0);
  71.     av_dict_set(&s->options, "out_sample_rate",    NULL, 0);
  72.  
  73.     return 0;
  74. }
  75.  
  76. static av_cold void uninit(AVFilterContext *ctx)
  77. {
  78.     ResampleContext *s = ctx->priv;
  79.  
  80.     if (s->avr) {
  81.         avresample_close(s->avr);
  82.         avresample_free(&s->avr);
  83.     }
  84.     av_dict_free(&s->options);
  85. }
  86.  
  87. static int query_formats(AVFilterContext *ctx)
  88. {
  89.     AVFilterLink *inlink  = ctx->inputs[0];
  90.     AVFilterLink *outlink = ctx->outputs[0];
  91.  
  92.     AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  93.     AVFilterFormats        *out_formats     = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  94.     AVFilterFormats        *in_samplerates  = ff_all_samplerates();
  95.     AVFilterFormats        *out_samplerates = ff_all_samplerates();
  96.     AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
  97.     AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();
  98.  
  99.     ff_formats_ref(in_formats,  &inlink->out_formats);
  100.     ff_formats_ref(out_formats, &outlink->in_formats);
  101.  
  102.     ff_formats_ref(in_samplerates,  &inlink->out_samplerates);
  103.     ff_formats_ref(out_samplerates, &outlink->in_samplerates);
  104.  
  105.     ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
  106.     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
  107.  
  108.     return 0;
  109. }
  110.  
  111. static int config_output(AVFilterLink *outlink)
  112. {
  113.     AVFilterContext *ctx = outlink->src;
  114.     AVFilterLink *inlink = ctx->inputs[0];
  115.     ResampleContext   *s = ctx->priv;
  116.     char buf1[64], buf2[64];
  117.     int ret;
  118.  
  119.     if (s->avr) {
  120.         avresample_close(s->avr);
  121.         avresample_free(&s->avr);
  122.     }
  123.  
  124.     if (inlink->channel_layout == outlink->channel_layout &&
  125.         inlink->sample_rate    == outlink->sample_rate    &&
  126.         (inlink->format        == outlink->format ||
  127.         (av_get_channel_layout_nb_channels(inlink->channel_layout)  == 1 &&
  128.          av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
  129.          av_get_planar_sample_fmt(inlink->format) ==
  130.          av_get_planar_sample_fmt(outlink->format))))
  131.         return 0;
  132.  
  133.     if (!(s->avr = avresample_alloc_context()))
  134.         return AVERROR(ENOMEM);
  135.  
  136.     if (s->options) {
  137.         AVDictionaryEntry *e = NULL;
  138.         while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
  139.             av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
  140.  
  141.         av_opt_set_dict(s->avr, &s->options);
  142.     }
  143.  
  144.     av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
  145.     av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
  146.     av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
  147.     av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
  148.     av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
  149.     av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
  150.  
  151.     if ((ret = avresample_open(s->avr)) < 0)
  152.         return ret;
  153.  
  154.     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
  155.     s->next_pts        = AV_NOPTS_VALUE;
  156.  
  157.     av_get_channel_layout_string(buf1, sizeof(buf1),
  158.                                  -1, inlink ->channel_layout);
  159.     av_get_channel_layout_string(buf2, sizeof(buf2),
  160.                                  -1, outlink->channel_layout);
  161.     av_log(ctx, AV_LOG_VERBOSE,
  162.            "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
  163.            av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
  164.            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
  165.  
  166.     return 0;
  167. }
  168.  
  169. static int request_frame(AVFilterLink *outlink)
  170. {
  171.     AVFilterContext *ctx = outlink->src;
  172.     ResampleContext   *s = ctx->priv;
  173.     int ret = 0;
  174.  
  175.     s->got_output = 0;
  176.     while (ret >= 0 && !s->got_output)
  177.         ret = ff_request_frame(ctx->inputs[0]);
  178.  
  179.     /* flush the lavr delay buffer */
  180.     if (ret == AVERROR_EOF && s->avr) {
  181.         AVFrame *frame;
  182.         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
  183.                                         outlink->sample_rate,
  184.                                         ctx->inputs[0]->sample_rate,
  185.                                         AV_ROUND_UP);
  186.  
  187.         if (!nb_samples)
  188.             return ret;
  189.  
  190.         frame = ff_get_audio_buffer(outlink, nb_samples);
  191.         if (!frame)
  192.             return AVERROR(ENOMEM);
  193.  
  194.         ret = avresample_convert(s->avr, frame->extended_data,
  195.                                  frame->linesize[0], nb_samples,
  196.                                  NULL, 0, 0);
  197.         if (ret <= 0) {
  198.             av_frame_free(&frame);
  199.             return (ret == 0) ? AVERROR_EOF : ret;
  200.         }
  201.  
  202.         frame->pts = s->next_pts;
  203.         return ff_filter_frame(outlink, frame);
  204.     }
  205.     return ret;
  206. }
  207.  
  208. static int filter_frame(AVFilterLink *inlink, AVFrame *in)
  209. {
  210.     AVFilterContext  *ctx = inlink->dst;
  211.     ResampleContext    *s = ctx->priv;
  212.     AVFilterLink *outlink = ctx->outputs[0];
  213.     int ret;
  214.  
  215.     if (s->avr) {
  216.         AVFrame *out;
  217.         int delay, nb_samples;
  218.  
  219.         /* maximum possible samples lavr can output */
  220.         delay      = avresample_get_delay(s->avr);
  221.         nb_samples = av_rescale_rnd(in->nb_samples + delay,
  222.                                     outlink->sample_rate, inlink->sample_rate,
  223.                                     AV_ROUND_UP);
  224.  
  225.         out = ff_get_audio_buffer(outlink, nb_samples);
  226.         if (!out) {
  227.             ret = AVERROR(ENOMEM);
  228.             goto fail;
  229.         }
  230.  
  231.         ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
  232.                                  nb_samples, in->extended_data, in->linesize[0],
  233.                                  in->nb_samples);
  234.         if (ret <= 0) {
  235.             av_frame_free(&out);
  236.             if (ret < 0)
  237.                 goto fail;
  238.         }
  239.  
  240.         av_assert0(!avresample_available(s->avr));
  241.  
  242.         if (s->next_pts == AV_NOPTS_VALUE) {
  243.             if (in->pts == AV_NOPTS_VALUE) {
  244.                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
  245.                        "assuming 0.\n");
  246.                 s->next_pts = 0;
  247.             } else
  248.                 s->next_pts = av_rescale_q(in->pts, inlink->time_base,
  249.                                            outlink->time_base);
  250.         }
  251.  
  252.         if (ret > 0) {
  253.             out->nb_samples = ret;
  254.             if (in->pts != AV_NOPTS_VALUE) {
  255.                 out->pts = av_rescale_q(in->pts, inlink->time_base,
  256.                                             outlink->time_base) -
  257.                                av_rescale(delay, outlink->sample_rate,
  258.                                           inlink->sample_rate);
  259.             } else
  260.                 out->pts = s->next_pts;
  261.  
  262.             s->next_pts = out->pts + out->nb_samples;
  263.  
  264.             ret = ff_filter_frame(outlink, out);
  265.             s->got_output = 1;
  266.         }
  267.  
  268. fail:
  269.         av_frame_free(&in);
  270.     } else {
  271.         in->format = outlink->format;
  272.         ret = ff_filter_frame(outlink, in);
  273.         s->got_output = 1;
  274.     }
  275.  
  276.     return ret;
  277. }
  278.  
  279. static const AVClass *resample_child_class_next(const AVClass *prev)
  280. {
  281.     return prev ? NULL : avresample_get_class();
  282. }
  283.  
  284. static void *resample_child_next(void *obj, void *prev)
  285. {
  286.     ResampleContext *s = obj;
  287.     return prev ? NULL : s->avr;
  288. }
  289.  
  290. static const AVClass resample_class = {
  291.     .class_name       = "resample",
  292.     .item_name        = av_default_item_name,
  293.     .version          = LIBAVUTIL_VERSION_INT,
  294.     .child_class_next = resample_child_class_next,
  295.     .child_next       = resample_child_next,
  296. };
  297.  
  298. static const AVFilterPad avfilter_af_resample_inputs[] = {
  299.     {
  300.         .name          = "default",
  301.         .type          = AVMEDIA_TYPE_AUDIO,
  302.         .filter_frame  = filter_frame,
  303.     },
  304.     { NULL }
  305. };
  306.  
  307. static const AVFilterPad avfilter_af_resample_outputs[] = {
  308.     {
  309.         .name          = "default",
  310.         .type          = AVMEDIA_TYPE_AUDIO,
  311.         .config_props  = config_output,
  312.         .request_frame = request_frame
  313.     },
  314.     { NULL }
  315. };
  316.  
  317. AVFilter avfilter_af_resample = {
  318.     .name          = "resample",
  319.     .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
  320.     .priv_size     = sizeof(ResampleContext),
  321.     .priv_class    = &resample_class,
  322.     .init_dict     = init,
  323.     .uninit        = uninit,
  324.     .query_formats = query_formats,
  325.     .inputs        = avfilter_af_resample_inputs,
  326.     .outputs       = avfilter_af_resample_outputs,
  327. };
  328.