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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. /**
  23.  * @file
  24.  * resampling audio filter
  25.  */
  26.  
  27. #include "libavutil/avstring.h"
  28. #include "libavutil/channel_layout.h"
  29. #include "libavutil/opt.h"
  30. #include "libavutil/samplefmt.h"
  31. #include "libavutil/avassert.h"
  32. #include "libswresample/swresample.h"
  33. #include "avfilter.h"
  34. #include "audio.h"
  35. #include "internal.h"
  36.  
  37. typedef struct {
  38.     const AVClass *class;
  39.     int sample_rate_arg;
  40.     double ratio;
  41.     struct SwrContext *swr;
  42.     int64_t next_pts;
  43.     int req_fullfilled;
  44.     int more_data;
  45. } AResampleContext;
  46.  
  47. static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
  48. {
  49.     AResampleContext *aresample = ctx->priv;
  50.     int ret = 0;
  51.  
  52.     aresample->next_pts = AV_NOPTS_VALUE;
  53.     aresample->swr = swr_alloc();
  54.     if (!aresample->swr) {
  55.         ret = AVERROR(ENOMEM);
  56.         goto end;
  57.     }
  58.  
  59.     if (opts) {
  60.         AVDictionaryEntry *e = NULL;
  61.  
  62.         while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
  63.             if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
  64.                 goto end;
  65.         }
  66.         av_dict_free(opts);
  67.     }
  68.     if (aresample->sample_rate_arg > 0)
  69.         av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
  70. end:
  71.     return ret;
  72. }
  73.  
  74. static av_cold void uninit(AVFilterContext *ctx)
  75. {
  76.     AResampleContext *aresample = ctx->priv;
  77.     swr_free(&aresample->swr);
  78. }
  79.  
  80. static int query_formats(AVFilterContext *ctx)
  81. {
  82.     AResampleContext *aresample = ctx->priv;
  83.     enum AVSampleFormat out_format;
  84.     int64_t out_rate, out_layout;
  85.  
  86.     AVFilterLink *inlink  = ctx->inputs[0];
  87.     AVFilterLink *outlink = ctx->outputs[0];
  88.  
  89.     AVFilterFormats        *in_formats, *out_formats;
  90.     AVFilterFormats        *in_samplerates, *out_samplerates;
  91.     AVFilterChannelLayouts *in_layouts, *out_layouts;
  92.  
  93.     av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
  94.     av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
  95.     av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
  96.  
  97.     in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  98.     if (!in_formats)
  99.         return AVERROR(ENOMEM);
  100.     ff_formats_ref  (in_formats,      &inlink->out_formats);
  101.  
  102.     in_samplerates  = ff_all_samplerates();
  103.     if (!in_samplerates)
  104.         return AVERROR(ENOMEM);
  105.     ff_formats_ref  (in_samplerates,  &inlink->out_samplerates);
  106.  
  107.     in_layouts      = ff_all_channel_counts();
  108.     if (!in_layouts)
  109.          return AVERROR(ENOMEM);
  110.     ff_channel_layouts_ref(in_layouts,      &inlink->out_channel_layouts);
  111.  
  112.     if(out_rate > 0) {
  113.         int ratelist[] = { out_rate, -1 };
  114.         out_samplerates = ff_make_format_list(ratelist);
  115.     } else {
  116.         out_samplerates = ff_all_samplerates();
  117.     }
  118.     if (!out_samplerates) {
  119.         av_log(ctx, AV_LOG_ERROR, "Cannot allocate output samplerates.\n");
  120.         return AVERROR(ENOMEM);
  121.     }
  122.  
  123.     ff_formats_ref(out_samplerates, &outlink->in_samplerates);
  124.  
  125.     if(out_format != AV_SAMPLE_FMT_NONE) {
  126.         int formatlist[] = { out_format, -1 };
  127.         out_formats = ff_make_format_list(formatlist);
  128.     } else
  129.         out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  130.     ff_formats_ref(out_formats, &outlink->in_formats);
  131.  
  132.     if(out_layout) {
  133.         int64_t layout_list[] = { out_layout, -1 };
  134.         out_layouts = avfilter_make_format64_list(layout_list);
  135.     } else
  136.         out_layouts = ff_all_channel_counts();
  137.     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
  138.  
  139.     return 0;
  140. }
  141.  
  142.  
  143. static int config_output(AVFilterLink *outlink)
  144. {
  145.     int ret;
  146.     AVFilterContext *ctx = outlink->src;
  147.     AVFilterLink *inlink = ctx->inputs[0];
  148.     AResampleContext *aresample = ctx->priv;
  149.     int64_t out_rate, out_layout;
  150.     enum AVSampleFormat out_format;
  151.     char inchl_buf[128], outchl_buf[128];
  152.  
  153.     aresample->swr = swr_alloc_set_opts(aresample->swr,
  154.                                         outlink->channel_layout, outlink->format, outlink->sample_rate,
  155.                                         inlink->channel_layout, inlink->format, inlink->sample_rate,
  156.                                         0, ctx);
  157.     if (!aresample->swr)
  158.         return AVERROR(ENOMEM);
  159.     if (!inlink->channel_layout)
  160.         av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
  161.     if (!outlink->channel_layout)
  162.         av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
  163.  
  164.     ret = swr_init(aresample->swr);
  165.     if (ret < 0)
  166.         return ret;
  167.  
  168.     av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
  169.     av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
  170.     av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
  171.     outlink->time_base = (AVRational) {1, out_rate};
  172.  
  173.     av_assert0(outlink->sample_rate == out_rate);
  174.     av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
  175.     av_assert0(outlink->format == out_format);
  176.  
  177.     aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
  178.  
  179.     av_get_channel_layout_string(inchl_buf,  sizeof(inchl_buf),  inlink ->channels, inlink ->channel_layout);
  180.     av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
  181.  
  182.     av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
  183.            inlink ->channels, inchl_buf,  av_get_sample_fmt_name(inlink->format),  inlink->sample_rate,
  184.            outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
  185.     return 0;
  186. }
  187.  
  188. static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
  189. {
  190.     AResampleContext *aresample = inlink->dst->priv;
  191.     const int n_in  = insamplesref->nb_samples;
  192.     int64_t delay;
  193.     int n_out       = n_in * aresample->ratio + 32;
  194.     AVFilterLink *const outlink = inlink->dst->outputs[0];
  195.     AVFrame *outsamplesref;
  196.     int ret;
  197.  
  198.     delay = swr_get_delay(aresample->swr, outlink->sample_rate);
  199.     if (delay > 0)
  200.         n_out += FFMIN(delay, FFMAX(4096, n_out));
  201.  
  202.     outsamplesref = ff_get_audio_buffer(outlink, n_out);
  203.  
  204.     if(!outsamplesref)
  205.         return AVERROR(ENOMEM);
  206.  
  207.     av_frame_copy_props(outsamplesref, insamplesref);
  208.     outsamplesref->format                = outlink->format;
  209.     av_frame_set_channels(outsamplesref, outlink->channels);
  210.     outsamplesref->channel_layout        = outlink->channel_layout;
  211.     outsamplesref->sample_rate           = outlink->sample_rate;
  212.  
  213.     if(insamplesref->pts != AV_NOPTS_VALUE) {
  214.         int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
  215.         int64_t outpts= swr_next_pts(aresample->swr, inpts);
  216.         aresample->next_pts =
  217.         outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
  218.     } else {
  219.         outsamplesref->pts  = AV_NOPTS_VALUE;
  220.     }
  221.     n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
  222.                                  (void *)insamplesref->extended_data, n_in);
  223.     if (n_out <= 0) {
  224.         av_frame_free(&outsamplesref);
  225.         av_frame_free(&insamplesref);
  226.         return 0;
  227.     }
  228.  
  229.     aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
  230.  
  231.     outsamplesref->nb_samples  = n_out;
  232.  
  233.     ret = ff_filter_frame(outlink, outsamplesref);
  234.     aresample->req_fullfilled= 1;
  235.     av_frame_free(&insamplesref);
  236.     return ret;
  237. }
  238.  
  239. static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
  240. {
  241.     AVFilterContext *ctx = outlink->src;
  242.     AResampleContext *aresample = ctx->priv;
  243.     AVFilterLink *const inlink = outlink->src->inputs[0];
  244.     AVFrame *outsamplesref;
  245.     int n_out = 4096;
  246.     int64_t pts;
  247.  
  248.     outsamplesref = ff_get_audio_buffer(outlink, n_out);
  249.     *outsamplesref_ret = outsamplesref;
  250.     if (!outsamplesref)
  251.         return AVERROR(ENOMEM);
  252.  
  253.     pts = swr_next_pts(aresample->swr, INT64_MIN);
  254.     pts = ROUNDED_DIV(pts, inlink->sample_rate);
  255.  
  256.     n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
  257.     if (n_out <= 0) {
  258.         av_frame_free(&outsamplesref);
  259.         return (n_out == 0) ? AVERROR_EOF : n_out;
  260.     }
  261.  
  262.     outsamplesref->sample_rate = outlink->sample_rate;
  263.     outsamplesref->nb_samples  = n_out;
  264.  
  265.     outsamplesref->pts = pts;
  266.  
  267.     return 0;
  268. }
  269.  
  270. static int request_frame(AVFilterLink *outlink)
  271. {
  272.     AVFilterContext *ctx = outlink->src;
  273.     AResampleContext *aresample = ctx->priv;
  274.     int ret;
  275.  
  276.     // First try to get data from the internal buffers
  277.     if (aresample->more_data) {
  278.         AVFrame *outsamplesref;
  279.  
  280.         if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
  281.             return ff_filter_frame(outlink, outsamplesref);
  282.         }
  283.     }
  284.     aresample->more_data = 0;
  285.  
  286.     // Second request more data from the input
  287.     aresample->req_fullfilled = 0;
  288.     do{
  289.         ret = ff_request_frame(ctx->inputs[0]);
  290.     }while(!aresample->req_fullfilled && ret>=0);
  291.  
  292.     // Third if we hit the end flush
  293.     if (ret == AVERROR_EOF) {
  294.         AVFrame *outsamplesref;
  295.  
  296.         if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
  297.             return ret;
  298.  
  299.         return ff_filter_frame(outlink, outsamplesref);
  300.     }
  301.     return ret;
  302. }
  303.  
  304. static const AVClass *resample_child_class_next(const AVClass *prev)
  305. {
  306.     return prev ? NULL : swr_get_class();
  307. }
  308.  
  309. static void *resample_child_next(void *obj, void *prev)
  310. {
  311.     AResampleContext *s = obj;
  312.     return prev ? NULL : s->swr;
  313. }
  314.  
  315. #define OFFSET(x) offsetof(AResampleContext, x)
  316. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  317.  
  318. static const AVOption options[] = {
  319.     {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0},  0,        INT_MAX, FLAGS },
  320.     {NULL}
  321. };
  322.  
  323. static const AVClass aresample_class = {
  324.     .class_name       = "aresample",
  325.     .item_name        = av_default_item_name,
  326.     .option           = options,
  327.     .version          = LIBAVUTIL_VERSION_INT,
  328.     .child_class_next = resample_child_class_next,
  329.     .child_next       = resample_child_next,
  330. };
  331.  
  332. static const AVFilterPad aresample_inputs[] = {
  333.     {
  334.         .name         = "default",
  335.         .type         = AVMEDIA_TYPE_AUDIO,
  336.         .filter_frame = filter_frame,
  337.     },
  338.     { NULL }
  339. };
  340.  
  341. static const AVFilterPad aresample_outputs[] = {
  342.     {
  343.         .name          = "default",
  344.         .config_props  = config_output,
  345.         .request_frame = request_frame,
  346.         .type          = AVMEDIA_TYPE_AUDIO,
  347.     },
  348.     { NULL }
  349. };
  350.  
  351. AVFilter ff_af_aresample = {
  352.     .name          = "aresample",
  353.     .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
  354.     .init_dict     = init_dict,
  355.     .uninit        = uninit,
  356.     .query_formats = query_formats,
  357.     .priv_size     = sizeof(AResampleContext),
  358.     .priv_class    = &aresample_class,
  359.     .inputs        = aresample_inputs,
  360.     .outputs       = aresample_outputs,
  361. };
  362.