git.sesse.net Git - ffmpeg/blob - libavfilter/af_aresample.c

   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/opt.h"
  29 #include "libavutil/samplefmt.h"
  30 #include "libavutil/avassert.h"
  31 #include "libswresample/swresample.h"
  32 #include "avfilter.h"
  33 #include "audio.h"
  34 #include "internal.h"
  35
  36 typedef struct {
  37     double ratio;
  38     struct SwrContext *swr;
  39     int64_t next_pts;
  40 } AResampleContext;
  41
  42 static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
  43 {
  44     AResampleContext *aresample = ctx->priv;
  45     int ret = 0;
  46     char *argd = av_strdup(args);
  47
  48     aresample->next_pts = AV_NOPTS_VALUE;
  49     aresample->swr = swr_alloc();
  50     if (!aresample->swr)
  51         return AVERROR(ENOMEM);
  52
  53     if (args) {
  54         char *ptr=argd, *token;
  55
  56         while(token = av_strtok(ptr, ":", &ptr)) {
  57             char *value;
  58             av_strtok(token, "=", &value);
  59
  60             if(value) {
  61                 if((ret=av_opt_set(aresample->swr, token, value, 0)) < 0)
  62                     goto end;
  63             } else {
  64                 int out_rate;
  65                 if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
  66                     goto end;
  67                 if((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
  68                     goto end;
  69             }
  70         }
  71     }
  72 end:
  73     av_free(argd);
  74     return ret;
  75 }
  76
  77 static av_cold void uninit(AVFilterContext *ctx)
  78 {
  79     AResampleContext *aresample = ctx->priv;
  80     swr_free(&aresample->swr);
  81 }
  82
  83 static int query_formats(AVFilterContext *ctx)
  84 {
  85     AResampleContext *aresample = ctx->priv;
  86     int out_rate                   = av_get_int(aresample->swr, "osr", NULL);
  87     uint64_t out_layout            = av_get_int(aresample->swr, "ocl", NULL);
  88     enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
  89
  90     AVFilterLink *inlink  = ctx->inputs[0];
  91     AVFilterLink *outlink = ctx->outputs[0];
  92
  93     AVFilterFormats        *in_formats      = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
  94     AVFilterFormats        *out_formats;
  95     AVFilterFormats        *in_samplerates  = ff_all_samplerates();
  96     AVFilterFormats        *out_samplerates;
  97     AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
  98     AVFilterChannelLayouts *out_layouts;
  99
 100     avfilter_formats_ref  (in_formats,      &inlink->out_formats);
 101     avfilter_formats_ref  (in_samplerates,  &inlink->out_samplerates);
 102     ff_channel_layouts_ref(in_layouts,      &inlink->out_channel_layouts);
 103
 104     if(out_rate > 0) {
 105         out_samplerates = avfilter_make_format_list((int[]){ out_rate, -1 });
 106     } else {
 107         out_samplerates = ff_all_samplerates();
 108     }
 109     avfilter_formats_ref(out_samplerates, &outlink->in_samplerates);
 110
 111     if(out_format != AV_SAMPLE_FMT_NONE) {
 112         out_formats = avfilter_make_format_list((int[]){ out_format, -1 });
 113     } else
 114         out_formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
 115     avfilter_formats_ref(out_formats, &outlink->in_formats);
 116
 117     if(out_layout) {
 118         out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
 119     } else
 120         out_layouts = ff_all_channel_layouts();
 121     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
 122
 123     return 0;
 124 }
 125
 126
 127 static int config_output(AVFilterLink *outlink)
 128 {
 129     int ret;
 130     AVFilterContext *ctx = outlink->src;
 131     AVFilterLink *inlink = ctx->inputs[0];
 132     AResampleContext *aresample = ctx->priv;
 133     int out_rate;
 134     uint64_t out_layout;
 135     enum AVSampleFormat out_format;
 136
 137     aresample->swr = swr_alloc_set_opts(aresample->swr,
 138                                         outlink->channel_layout, outlink->format, outlink->sample_rate,
 139                                         inlink->channel_layout, inlink->format, inlink->sample_rate,
 140                                         0, ctx);
 141     if (!aresample->swr)
 142         return AVERROR(ENOMEM);
 143
 144     ret = swr_init(aresample->swr);
 145     if (ret < 0)
 146         return ret;
 147
 148     out_rate   = av_get_int(aresample->swr, "osr", NULL);
 149     out_layout = av_get_int(aresample->swr, "ocl", NULL);
 150     out_format = av_get_int(aresample->swr, "osf", NULL);
 151     outlink->time_base = (AVRational) {1, out_rate};
 152
 153     av_assert0(outlink->sample_rate == out_rate);
 154     av_assert0(outlink->channel_layout == out_layout);
 155     av_assert0(outlink->format == out_format);
 156
 157     aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
 158
 159     av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
 160            inlink->sample_rate, outlink->sample_rate);
 161     return 0;
 162 }
 163
 164 static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
 165 {
 166     AResampleContext *aresample = inlink->dst->priv;
 167     const int n_in  = insamplesref->audio->nb_samples;
 168     int n_out       = n_in * aresample->ratio;
 169     AVFilterLink *const outlink = inlink->dst->outputs[0];
 170     AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
 171
 172     n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
 173                                  (void *)insamplesref->data, n_in);
 174     if (n_out <= 0) {
 175         avfilter_unref_buffer(outsamplesref);
 176         avfilter_unref_buffer(insamplesref);
 177         return;
 178     }
 179
 180     avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
 181
 182     outsamplesref->audio->sample_rate = outlink->sample_rate;
 183     outsamplesref->audio->nb_samples  = n_out;
 184
 185     if(insamplesref->pts != AV_NOPTS_VALUE) {
 186         aresample->next_pts = insamplesref->pts;
 187         outsamplesref->pts  = av_rescale_q(insamplesref->pts, inlink->time_base, outlink->time_base);
 188     } else{
 189         outsamplesref->pts  = AV_NOPTS_VALUE; //aresample->next_pts;
 190     }
 191     if(aresample->next_pts != AV_NOPTS_VALUE)
 192         aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
 193
 194     ff_filter_samples(outlink, outsamplesref);
 195     avfilter_unref_buffer(insamplesref);
 196 }
 197
 198 static int request_frame(AVFilterLink *outlink)
 199 {
 200     AVFilterContext *ctx = outlink->src;
 201     AResampleContext *aresample = ctx->priv;
 202     int ret = avfilter_request_frame(ctx->inputs[0]);
 203
 204     if (ret == AVERROR_EOF) {
 205         AVFilterBufferRef *outsamplesref;
 206         int n_out = 4096;
 207
 208         outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
 209         if (!outsamplesref)
 210             return AVERROR(ENOMEM);
 211         n_out = swr_convert(aresample->swr, outsamplesref->data, n_out, 0, 0);
 212         if (n_out <= 0) {
 213             avfilter_unref_buffer(outsamplesref);
 214             return (n_out == 0) ? AVERROR_EOF : n_out;
 215         }
 216
 217         outsamplesref->audio->sample_rate = outlink->sample_rate;
 218         outsamplesref->audio->nb_samples  = n_out;
 219         outsamplesref->pts = aresample->next_pts;
 220         if(aresample->next_pts != AV_NOPTS_VALUE)
 221             aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
 222
 223         ff_filter_samples(outlink, outsamplesref);
 224         return 0;
 225     }
 226     return ret;
 227 }
 228
 229 AVFilter avfilter_af_aresample = {
 230     .name          = "aresample",
 231     .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
 232     .init          = init,
 233     .uninit        = uninit,
 234     .query_formats = query_formats,
 235     .priv_size     = sizeof(AResampleContext),
 236
 237     .inputs    = (const AVFilterPad[]) {{ .name      = "default",
 238                                     .type            = AVMEDIA_TYPE_AUDIO,
 239                                     .filter_samples  = filter_samples,
 240                                     .min_perms       = AV_PERM_READ, },
 241                                   { .name = NULL}},
 242     .outputs   = (const AVFilterPad[]) {{ .name      = "default",
 243                                     .config_props    = config_output,
 244                                     .request_frame   = request_frame,
 245                                     .type            = AVMEDIA_TYPE_AUDIO, },
 246                                   { .name = NULL}},
 247 };