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 "internal.h"
  35
  36 typedef struct ResampleContext {
  37     AVAudioResampleContext *avr;
  38
  39     int64_t next_pts;
  40 } ResampleContext;
  41
  42 static av_cold void uninit(AVFilterContext *ctx)
  43 {
  44     ResampleContext *s = ctx->priv;
  45
  46     if (s->avr) {
  47         avresample_close(s->avr);
  48         avresample_free(&s->avr);
  49     }
  50 }
  51
  52 static int query_formats(AVFilterContext *ctx)
  53 {
  54     AVFilterLink *inlink  = ctx->inputs[0];
  55     AVFilterLink *outlink = ctx->outputs[0];
  56
  57     AVFilterFormats        *in_formats      = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
  58     AVFilterFormats        *out_formats     = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
  59
  60     avfilter_formats_ref(in_formats,  &inlink->out_formats);
  61     avfilter_formats_ref(out_formats, &outlink->in_formats);
  62
  63     return 0;
  64 }
  65
  66 static int config_output(AVFilterLink *outlink)
  67 {
  68     AVFilterContext *ctx = outlink->src;
  69     AVFilterLink *inlink = ctx->inputs[0];
  70     ResampleContext   *s = ctx->priv;
  71     char buf1[64], buf2[64];
  72     int ret;
  73
  74     if (s->avr) {
  75         avresample_close(s->avr);
  76         avresample_free(&s->avr);
  77     }
  78
  79     if (inlink->channel_layout == outlink->channel_layout &&
  80         inlink->sample_rate    == outlink->sample_rate    &&
  81         inlink->format         == outlink->format)
  82         return 0;
  83
  84     if (!(s->avr = avresample_alloc_context()))
  85         return AVERROR(ENOMEM);
  86
  87     av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
  88     av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
  89     av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
  90     av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
  91     av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
  92     av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
  93
  94     /* if both the input and output formats are s16 or u8, use s16 as
  95        the internal sample format */
  96     if (av_get_bytes_per_sample(inlink->format)  <= 2 &&
  97         av_get_bytes_per_sample(outlink->format) <= 2)
  98         av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
  99
 100     if ((ret = avresample_open(s->avr)) < 0)
 101         return ret;
 102
 103     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
 104     s->next_pts        = AV_NOPTS_VALUE;
 105
 106     av_get_channel_layout_string(buf1, sizeof(buf1),
 107                                  -1, inlink ->channel_layout);
 108     av_get_channel_layout_string(buf2, sizeof(buf2),
 109                                  -1, outlink->channel_layout);
 110     av_log(ctx, AV_LOG_VERBOSE,
 111            "fmt:%s srate: %d cl:%s -> fmt:%s srate: %d cl:%s\n",
 112            av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
 113            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
 114
 115     return 0;
 116 }
 117
 118 static int request_frame(AVFilterLink *outlink)
 119 {
 120     AVFilterContext *ctx = outlink->src;
 121     ResampleContext   *s = ctx->priv;
 122     int ret = avfilter_request_frame(ctx->inputs[0]);
 123
 124     /* flush the lavr delay buffer */
 125     if (ret == AVERROR_EOF && s->avr) {
 126         AVFilterBufferRef *buf;
 127         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
 128                                         outlink->sample_rate,
 129                                         ctx->inputs[0]->sample_rate,
 130                                         AV_ROUND_UP);
 131
 132         if (!nb_samples)
 133             return ret;
 134
 135         buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
 136         if (!buf)
 137             return AVERROR(ENOMEM);
 138
 139         ret = avresample_convert(s->avr, (void**)buf->extended_data,
 140                                  buf->linesize[0], nb_samples,
 141                                  NULL, 0, 0);
 142         if (ret <= 0) {
 143             avfilter_unref_buffer(buf);
 144             return (ret == 0) ? AVERROR_EOF : ret;
 145         }
 146
 147         buf->pts = s->next_pts;
 148         ff_filter_samples(outlink, buf);
 149         return 0;
 150     }
 151     return ret;
 152 }
 153
 154 static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 155 {
 156     AVFilterContext  *ctx = inlink->dst;
 157     ResampleContext    *s = ctx->priv;
 158     AVFilterLink *outlink = ctx->outputs[0];
 159
 160     if (s->avr) {
 161         AVFilterBufferRef *buf_out;
 162         int delay, nb_samples, ret;
 163
 164         /* maximum possible samples lavr can output */
 165         delay      = avresample_get_delay(s->avr);
 166         nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
 167                                     outlink->sample_rate, inlink->sample_rate,
 168                                     AV_ROUND_UP);
 169
 170         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
 171         ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
 172                                      buf_out->linesize[0], nb_samples,
 173                                      (void**)buf->extended_data, buf->linesize[0],
 174                                      buf->audio->nb_samples);
 175
 176         av_assert0(!avresample_available(s->avr));
 177
 178         if (s->next_pts == AV_NOPTS_VALUE) {
 179             if (buf->pts == AV_NOPTS_VALUE) {
 180                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
 181                        "assuming 0.\n");
 182                 s->next_pts = 0;
 183             } else
 184                 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
 185                                            outlink->time_base);
 186         }
 187
 188         if (ret > 0) {
 189             buf_out->audio->nb_samples = ret;
 190             if (buf->pts != AV_NOPTS_VALUE) {
 191                 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
 192                                             outlink->time_base) -
 193                                av_rescale(delay, outlink->sample_rate,
 194                                           inlink->sample_rate);
 195             } else
 196                 buf_out->pts = s->next_pts;
 197
 198             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
 199
 200             ff_filter_samples(outlink, buf_out);
 201         }
 202         avfilter_unref_buffer(buf);
 203     } else
 204         ff_filter_samples(outlink, buf);
 205 }
 206
 207 AVFilter avfilter_af_resample = {
 208     .name          = "resample",
 209     .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
 210     .priv_size     = sizeof(ResampleContext),
 211
 212     .uninit         = uninit,
 213     .query_formats  = query_formats,
 214
 215     .inputs    = (const AVFilterPad[]) {{ .name            = "default",
 216                                           .type            = AVMEDIA_TYPE_AUDIO,
 217                                           .filter_samples  = filter_samples,
 218                                           .min_perms       = AV_PERM_READ },
 219                                         { .name = NULL}},
 220     .outputs   = (const AVFilterPad[]) {{ .name          = "default",
 221                                           .type          = AVMEDIA_TYPE_AUDIO,
 222                                           .config_props  = config_output,
 223                                           .request_frame = request_frame },
 224                                         { .name = NULL}},
 225 };