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

   1 /*
   2  *
   3  * This file is part of Libav.
   4  *
   5  * Libav is free software; you can redistribute it and/or
   6  * modify it under the terms of the GNU Lesser General Public
   7  * License as published by the Free Software Foundation; either
   8  * version 2.1 of the License, or (at your option) any later version.
   9  *
  10  * Libav is distributed in the hope that it will be useful,
  11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  13  * Lesser General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU Lesser General Public
  16  * License along with Libav; if not, write to the Free Software
  17  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  18  */
  19
  20 /**
  21  * @file
  22  * sample format and channel layout conversion audio filter
  23  */
  24
  25 #include "libavutil/avassert.h"
  26 #include "libavutil/avstring.h"
  27 #include "libavutil/mathematics.h"
  28 #include "libavutil/opt.h"
  29
  30 #include "libavresample/avresample.h"
  31
  32 #include "audio.h"
  33 #include "avfilter.h"
  34 #include "formats.h"
  35 #include "internal.h"
  36
  37 typedef struct ResampleContext {
  38     AVAudioResampleContext *avr;
  39
  40     int64_t next_pts;
  41
  42     /* set by filter_samples() to signal an output frame to request_frame() */
  43     int got_output;
  44 } ResampleContext;
  45
  46 static av_cold void uninit(AVFilterContext *ctx)
  47 {
  48     ResampleContext *s = ctx->priv;
  49
  50     if (s->avr) {
  51         avresample_close(s->avr);
  52         avresample_free(&s->avr);
  53     }
  54 }
  55
  56 static int query_formats(AVFilterContext *ctx)
  57 {
  58     AVFilterLink *inlink  = ctx->inputs[0];
  59     AVFilterLink *outlink = ctx->outputs[0];
  60
  61     AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  62     AVFilterFormats        *out_formats     = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  63     AVFilterFormats        *in_samplerates  = ff_all_samplerates();
  64     AVFilterFormats        *out_samplerates = ff_all_samplerates();
  65     AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
  66     AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();
  67
  68     ff_formats_ref(in_formats,  &inlink->out_formats);
  69     ff_formats_ref(out_formats, &outlink->in_formats);
  70
  71     ff_formats_ref(in_samplerates,  &inlink->out_samplerates);
  72     ff_formats_ref(out_samplerates, &outlink->in_samplerates);
  73
  74     ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
  75     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
  76
  77     return 0;
  78 }
  79
  80 static int config_output(AVFilterLink *outlink)
  81 {
  82     AVFilterContext *ctx = outlink->src;
  83     AVFilterLink *inlink = ctx->inputs[0];
  84     ResampleContext   *s = ctx->priv;
  85     char buf1[64], buf2[64];
  86     int ret;
  87
  88     if (s->avr) {
  89         avresample_close(s->avr);
  90         avresample_free(&s->avr);
  91     }
  92
  93     if (inlink->channel_layout == outlink->channel_layout &&
  94         inlink->sample_rate    == outlink->sample_rate    &&
  95         inlink->format         == outlink->format)
  96         return 0;
  97
  98     if (!(s->avr = avresample_alloc_context()))
  99         return AVERROR(ENOMEM);
 100
 101     av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
 102     av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
 103     av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
 104     av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
 105     av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
 106     av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
 107
 108     if ((ret = avresample_open(s->avr)) < 0)
 109         return ret;
 110
 111     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
 112     s->next_pts        = AV_NOPTS_VALUE;
 113
 114     av_get_channel_layout_string(buf1, sizeof(buf1),
 115                                  -1, inlink ->channel_layout);
 116     av_get_channel_layout_string(buf2, sizeof(buf2),
 117                                  -1, outlink->channel_layout);
 118     av_log(ctx, AV_LOG_VERBOSE,
 119            "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
 120            av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
 121            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
 122
 123     return 0;
 124 }
 125
 126 static int request_frame(AVFilterLink *outlink)
 127 {
 128     AVFilterContext *ctx = outlink->src;
 129     ResampleContext   *s = ctx->priv;
 130     int ret = 0;
 131
 132     s->got_output = 0;
 133     while (ret >= 0 && !s->got_output)
 134         ret = ff_request_frame(ctx->inputs[0]);
 135
 136     /* flush the lavr delay buffer */
 137     if (ret == AVERROR_EOF && s->avr) {
 138         AVFilterBufferRef *buf;
 139         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
 140                                         outlink->sample_rate,
 141                                         ctx->inputs[0]->sample_rate,
 142                                         AV_ROUND_UP);
 143
 144         if (!nb_samples)
 145             return ret;
 146
 147         buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
 148         if (!buf)
 149             return AVERROR(ENOMEM);
 150
 151         ret = avresample_convert(s->avr, (void**)buf->extended_data,
 152                                  buf->linesize[0], nb_samples,
 153                                  NULL, 0, 0);
 154         if (ret <= 0) {
 155             avfilter_unref_buffer(buf);
 156             return (ret == 0) ? AVERROR_EOF : ret;
 157         }
 158
 159         buf->pts = s->next_pts;
 160         return ff_filter_samples(outlink, buf);
 161     }
 162     return ret;
 163 }
 164
 165 static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 166 {
 167     AVFilterContext  *ctx = inlink->dst;
 168     ResampleContext    *s = ctx->priv;
 169     AVFilterLink *outlink = ctx->outputs[0];
 170     int ret;
 171
 172     if (s->avr) {
 173         AVFilterBufferRef *buf_out;
 174         int delay, nb_samples;
 175
 176         /* maximum possible samples lavr can output */
 177         delay      = avresample_get_delay(s->avr);
 178         nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
 179                                     outlink->sample_rate, inlink->sample_rate,
 180                                     AV_ROUND_UP);
 181
 182         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
 183         if (!buf_out) {
 184             ret = AVERROR(ENOMEM);
 185             goto fail;
 186         }
 187
 188         ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
 189                                      buf_out->linesize[0], nb_samples,
 190                                      (void**)buf->extended_data, buf->linesize[0],
 191                                      buf->audio->nb_samples);
 192         if (ret < 0) {
 193             avfilter_unref_buffer(buf_out);
 194             goto fail;
 195         }
 196
 197         av_assert0(!avresample_available(s->avr));
 198
 199         if (s->next_pts == AV_NOPTS_VALUE) {
 200             if (buf->pts == AV_NOPTS_VALUE) {
 201                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
 202                        "assuming 0.\n");
 203                 s->next_pts = 0;
 204             } else
 205                 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
 206                                            outlink->time_base);
 207         }
 208
 209         if (ret > 0) {
 210             buf_out->audio->nb_samples = ret;
 211             if (buf->pts != AV_NOPTS_VALUE) {
 212                 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
 213                                             outlink->time_base) -
 214                                av_rescale(delay, outlink->sample_rate,
 215                                           inlink->sample_rate);
 216             } else
 217                 buf_out->pts = s->next_pts;
 218
 219             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
 220
 221             ret = ff_filter_samples(outlink, buf_out);
 222             s->got_output = 1;
 223         }
 224
 225 fail:
 226         avfilter_unref_buffer(buf);
 227     } else {
 228         ret = ff_filter_samples(outlink, buf);
 229         s->got_output = 1;
 230     }
 231
 232     return ret;
 233 }
 234
 235 AVFilter avfilter_af_resample = {
 236     .name          = "resample",
 237     .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
 238     .priv_size     = sizeof(ResampleContext),
 239
 240     .uninit         = uninit,
 241     .query_formats  = query_formats,
 242
 243     .inputs    = (const AVFilterPad[]) {{ .name            = "default",
 244                                           .type            = AVMEDIA_TYPE_AUDIO,
 245                                           .filter_samples  = filter_samples,
 246                                           .min_perms       = AV_PERM_READ },
 247                                         { .name = NULL}},
 248     .outputs   = (const AVFilterPad[]) {{ .name          = "default",
 249                                           .type          = AVMEDIA_TYPE_AUDIO,
 250                                           .config_props  = config_output,
 251                                           .request_frame = request_frame },
 252                                         { .name = NULL}},
 253 };