LCOV - code coverage report
Current view: top level - src/libavfilter - af_aresample.c (source / functions) Hit Total Coverage
Test: coverage.info Lines: 132 154 85.7 %
Date: 2017-01-24 04:42:20 Functions: 8 9 88.9 %

          Line data    Source code
       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 more_data;
      44             : } AResampleContext;
      45             : 
      46        1060 : static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
      47             : {
      48        1060 :     AResampleContext *aresample = ctx->priv;
      49        1060 :     int ret = 0;
      50             : 
      51        1060 :     aresample->next_pts = AV_NOPTS_VALUE;
      52        1060 :     aresample->swr = swr_alloc();
      53        1060 :     if (!aresample->swr) {
      54           0 :         ret = AVERROR(ENOMEM);
      55           0 :         goto end;
      56             :     }
      57             : 
      58        1060 :     if (opts) {
      59        1060 :         AVDictionaryEntry *e = NULL;
      60             : 
      61        3275 :         while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
      62        1155 :             if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
      63           0 :                 goto end;
      64             :         }
      65        1060 :         av_dict_free(opts);
      66             :     }
      67        1060 :     if (aresample->sample_rate_arg > 0)
      68         516 :         av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
      69             : end:
      70        1060 :     return ret;
      71             : }
      72             : 
      73        1060 : static av_cold void uninit(AVFilterContext *ctx)
      74             : {
      75        1060 :     AResampleContext *aresample = ctx->priv;
      76        1060 :     swr_free(&aresample->swr);
      77        1060 : }
      78             : 
      79        1060 : static int query_formats(AVFilterContext *ctx)
      80             : {
      81        1060 :     AResampleContext *aresample = ctx->priv;
      82             :     enum AVSampleFormat out_format;
      83             :     int64_t out_rate, out_layout;
      84             : 
      85        1060 :     AVFilterLink *inlink  = ctx->inputs[0];
      86        1060 :     AVFilterLink *outlink = ctx->outputs[0];
      87             : 
      88             :     AVFilterFormats        *in_formats, *out_formats;
      89             :     AVFilterFormats        *in_samplerates, *out_samplerates;
      90             :     AVFilterChannelLayouts *in_layouts, *out_layouts;
      91             :     int ret;
      92             : 
      93        1060 :     av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
      94        1060 :     av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
      95        1060 :     av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
      96             : 
      97        1060 :     in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
      98        1060 :     if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0)
      99           0 :         return ret;
     100             : 
     101        1060 :     in_samplerates  = ff_all_samplerates();
     102        1060 :     if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0)
     103           0 :         return ret;
     104             : 
     105        1060 :     in_layouts      = ff_all_channel_counts();
     106        1060 :     if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0)
     107           0 :         return ret;
     108             : 
     109        1060 :     if(out_rate > 0) {
     110         516 :         int ratelist[] = { out_rate, -1 };
     111         516 :         out_samplerates = ff_make_format_list(ratelist);
     112             :     } else {
     113         544 :         out_samplerates = ff_all_samplerates();
     114             :     }
     115             : 
     116        1060 :     if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0)
     117           0 :         return ret;
     118             : 
     119        1060 :     if(out_format != AV_SAMPLE_FMT_NONE) {
     120           0 :         int formatlist[] = { out_format, -1 };
     121           0 :         out_formats = ff_make_format_list(formatlist);
     122             :     } else
     123        1060 :         out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
     124        1060 :     if ((ret = ff_formats_ref(out_formats, &outlink->in_formats)) < 0)
     125           0 :         return ret;
     126             : 
     127        1060 :     if(out_layout) {
     128           0 :         int64_t layout_list[] = { out_layout, -1 };
     129           0 :         out_layouts = avfilter_make_format64_list(layout_list);
     130             :     } else
     131        1060 :         out_layouts = ff_all_channel_counts();
     132             : 
     133        1060 :     return ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
     134             : }
     135             : 
     136             : 
     137        1060 : static int config_output(AVFilterLink *outlink)
     138             : {
     139             :     int ret;
     140        1060 :     AVFilterContext *ctx = outlink->src;
     141        1060 :     AVFilterLink *inlink = ctx->inputs[0];
     142        1060 :     AResampleContext *aresample = ctx->priv;
     143             :     int64_t out_rate, out_layout;
     144             :     enum AVSampleFormat out_format;
     145             :     char inchl_buf[128], outchl_buf[128];
     146             : 
     147        3180 :     aresample->swr = swr_alloc_set_opts(aresample->swr,
     148        1060 :                                         outlink->channel_layout, outlink->format, outlink->sample_rate,
     149        1060 :                                         inlink->channel_layout, inlink->format, inlink->sample_rate,
     150             :                                         0, ctx);
     151        1060 :     if (!aresample->swr)
     152           0 :         return AVERROR(ENOMEM);
     153        1060 :     if (!inlink->channel_layout)
     154           3 :         av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
     155        1060 :     if (!outlink->channel_layout)
     156           2 :         av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
     157             : 
     158        1060 :     ret = swr_init(aresample->swr);
     159        1060 :     if (ret < 0)
     160           0 :         return ret;
     161             : 
     162        1060 :     av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
     163        1060 :     av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
     164        1060 :     av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
     165        1060 :     outlink->time_base = (AVRational) {1, out_rate};
     166             : 
     167        1060 :     av_assert0(outlink->sample_rate == out_rate);
     168        1060 :     av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
     169        1060 :     av_assert0(outlink->format == out_format);
     170             : 
     171        1060 :     aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
     172             : 
     173        1060 :     av_get_channel_layout_string(inchl_buf,  sizeof(inchl_buf),  inlink ->channels, inlink ->channel_layout);
     174        1060 :     av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
     175             : 
     176        3180 :     av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
     177        1060 :            inlink ->channels, inchl_buf,  av_get_sample_fmt_name(inlink->format),  inlink->sample_rate,
     178        1060 :            outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
     179        1060 :     return 0;
     180             : }
     181             : 
     182      233611 : static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
     183             : {
     184      233611 :     AResampleContext *aresample = inlink->dst->priv;
     185      233611 :     const int n_in  = insamplesref->nb_samples;
     186             :     int64_t delay;
     187      233611 :     int n_out       = n_in * aresample->ratio + 32;
     188      233611 :     AVFilterLink *const outlink = inlink->dst->outputs[0];
     189             :     AVFrame *outsamplesref;
     190             :     int ret;
     191             : 
     192      233611 :     delay = swr_get_delay(aresample->swr, outlink->sample_rate);
     193      233611 :     if (delay > 0)
     194        3749 :         n_out += FFMIN(delay, FFMAX(4096, n_out));
     195             : 
     196      233611 :     outsamplesref = ff_get_audio_buffer(outlink, n_out);
     197             : 
     198      233611 :     if(!outsamplesref)
     199           0 :         return AVERROR(ENOMEM);
     200             : 
     201      233611 :     av_frame_copy_props(outsamplesref, insamplesref);
     202      233611 :     outsamplesref->format                = outlink->format;
     203      233611 :     av_frame_set_channels(outsamplesref, outlink->channels);
     204      233611 :     outsamplesref->channel_layout        = outlink->channel_layout;
     205      233611 :     outsamplesref->sample_rate           = outlink->sample_rate;
     206             : 
     207      233611 :     if(insamplesref->pts != AV_NOPTS_VALUE) {
     208      233270 :         int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
     209      233270 :         int64_t outpts= swr_next_pts(aresample->swr, inpts);
     210      233270 :         aresample->next_pts =
     211      233270 :         outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
     212             :     } else {
     213         341 :         outsamplesref->pts  = AV_NOPTS_VALUE;
     214             :     }
     215      233611 :     n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
     216      233611 :                                  (void *)insamplesref->extended_data, n_in);
     217      233611 :     if (n_out <= 0) {
     218           0 :         av_frame_free(&outsamplesref);
     219           0 :         av_frame_free(&insamplesref);
     220           0 :         return 0;
     221             :     }
     222             : 
     223      233611 :     aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
     224             : 
     225      233611 :     outsamplesref->nb_samples  = n_out;
     226             : 
     227      233611 :     ret = ff_filter_frame(outlink, outsamplesref);
     228      233611 :     av_frame_free(&insamplesref);
     229      233611 :     return ret;
     230             : }
     231             : 
     232        1653 : static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
     233             : {
     234        1653 :     AVFilterContext *ctx = outlink->src;
     235        1653 :     AResampleContext *aresample = ctx->priv;
     236        1653 :     AVFilterLink *const inlink = outlink->src->inputs[0];
     237             :     AVFrame *outsamplesref;
     238        1653 :     int n_out = 4096;
     239             :     int64_t pts;
     240             : 
     241        1653 :     outsamplesref = ff_get_audio_buffer(outlink, n_out);
     242        1653 :     *outsamplesref_ret = outsamplesref;
     243        1653 :     if (!outsamplesref)
     244           0 :         return AVERROR(ENOMEM);
     245             : 
     246        1653 :     pts = swr_next_pts(aresample->swr, INT64_MIN);
     247        1653 :     pts = ROUNDED_DIV(pts, inlink->sample_rate);
     248             : 
     249        1653 :     n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
     250        1653 :     if (n_out <= 0) {
     251        1010 :         av_frame_free(&outsamplesref);
     252        1010 :         return (n_out == 0) ? AVERROR_EOF : n_out;
     253             :     }
     254             : 
     255         643 :     outsamplesref->sample_rate = outlink->sample_rate;
     256         643 :     outsamplesref->nb_samples  = n_out;
     257             : 
     258         643 :     outsamplesref->pts = pts;
     259             : 
     260         643 :     return 0;
     261             : }
     262             : 
     263      215053 : static int request_frame(AVFilterLink *outlink)
     264             : {
     265      215053 :     AVFilterContext *ctx = outlink->src;
     266      215053 :     AResampleContext *aresample = ctx->priv;
     267             :     int ret;
     268             : 
     269             :     // First try to get data from the internal buffers
     270      215053 :     if (aresample->more_data) {
     271             :         AVFrame *outsamplesref;
     272             : 
     273         123 :         if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
     274         121 :             return ff_filter_frame(outlink, outsamplesref);
     275             :         }
     276             :     }
     277      214932 :     aresample->more_data = 0;
     278             : 
     279             :     // Second request more data from the input
     280      214932 :     ret = ff_request_frame(ctx->inputs[0]);
     281             : 
     282             :     // Third if we hit the end flush
     283      214932 :     if (ret == AVERROR_EOF) {
     284             :         AVFrame *outsamplesref;
     285             : 
     286        1530 :         if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
     287        1008 :             return ret;
     288             : 
     289         522 :         return ff_filter_frame(outlink, outsamplesref);
     290             :     }
     291      213402 :     return ret;
     292             : }
     293             : 
     294        1155 : static const AVClass *resample_child_class_next(const AVClass *prev)
     295             : {
     296        1155 :     return prev ? NULL : swr_get_class();
     297             : }
     298             : 
     299           0 : static void *resample_child_next(void *obj, void *prev)
     300             : {
     301           0 :     AResampleContext *s = obj;
     302           0 :     return prev ? NULL : s->swr;
     303             : }
     304             : 
     305             : #define OFFSET(x) offsetof(AResampleContext, x)
     306             : #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
     307             : 
     308             : static const AVOption options[] = {
     309             :     {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0},  0,        INT_MAX, FLAGS },
     310             :     {NULL}
     311             : };
     312             : 
     313             : static const AVClass aresample_class = {
     314             :     .class_name       = "aresample",
     315             :     .item_name        = av_default_item_name,
     316             :     .option           = options,
     317             :     .version          = LIBAVUTIL_VERSION_INT,
     318             :     .child_class_next = resample_child_class_next,
     319             :     .child_next       = resample_child_next,
     320             : };
     321             : 
     322             : static const AVFilterPad aresample_inputs[] = {
     323             :     {
     324             :         .name         = "default",
     325             :         .type         = AVMEDIA_TYPE_AUDIO,
     326             :         .filter_frame = filter_frame,
     327             :     },
     328             :     { NULL }
     329             : };
     330             : 
     331             : static const AVFilterPad aresample_outputs[] = {
     332             :     {
     333             :         .name          = "default",
     334             :         .config_props  = config_output,
     335             :         .request_frame = request_frame,
     336             :         .type          = AVMEDIA_TYPE_AUDIO,
     337             :     },
     338             :     { NULL }
     339             : };
     340             : 
     341             : AVFilter ff_af_aresample = {
     342             :     .name          = "aresample",
     343             :     .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
     344             :     .init_dict     = init_dict,
     345             :     .uninit        = uninit,
     346             :     .query_formats = query_formats,
     347             :     .priv_size     = sizeof(AResampleContext),
     348             :     .priv_class    = &aresample_class,
     349             :     .inputs        = aresample_inputs,
     350             :     .outputs       = aresample_outputs,
     351             : };

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