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 } ResampleContext;
  42
  43 static av_cold void uninit(AVFilterContext *ctx)
  44 {
  45     ResampleContext *s = ctx->priv;
  46
  47     if (s->avr) {
  48         avresample_close(s->avr);
  49         avresample_free(&s->avr);
  50     }
  51 }
  52
  53 static int query_formats(AVFilterContext *ctx)
  54 {
  55     AVFilterLink *inlink  = ctx->inputs[0];
  56     AVFilterLink *outlink = ctx->outputs[0];
  57
  58     AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  59     AVFilterFormats        *out_formats     = ff_all_formats(AVMEDIA_TYPE_AUDIO);
  60     AVFilterFormats        *in_samplerates  = ff_all_samplerates();
  61     AVFilterFormats        *out_samplerates = ff_all_samplerates();
  62     AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
  63     AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();
  64
  65     ff_formats_ref(in_formats,  &inlink->out_formats);
  66     ff_formats_ref(out_formats, &outlink->in_formats);
  67
  68     ff_formats_ref(in_samplerates,  &inlink->out_samplerates);
  69     ff_formats_ref(out_samplerates, &outlink->in_samplerates);
  70
  71     ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
  72     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
  73
  74     return 0;
  75 }
  76
  77 static int config_output(AVFilterLink *outlink)
  78 {
  79     AVFilterContext *ctx = outlink->src;
  80     AVFilterLink *inlink = ctx->inputs[0];
  81     ResampleContext   *s = ctx->priv;
  82     char buf1[64], buf2[64];
  83     int ret;
  84
  85     if (s->avr) {
  86         avresample_close(s->avr);
  87         avresample_free(&s->avr);
  88     }
  89
  90     if (inlink->channel_layout == outlink->channel_layout &&
  91         inlink->sample_rate    == outlink->sample_rate    &&
  92         inlink->format         == outlink->format)
  93         return 0;
  94
  95     if (!(s->avr = avresample_alloc_context()))
  96         return AVERROR(ENOMEM);
  97
  98     av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
  99     av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
 100     av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
 101     av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
 102     av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
 103     av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
 104
 105     /* if both the input and output formats are s16 or u8, use s16 as
 106        the internal sample format */
 107     if (av_get_bytes_per_sample(inlink->format)  <= 2 &&
 108         av_get_bytes_per_sample(outlink->format) <= 2)
 109         av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
 110
 111     if ((ret = avresample_open(s->avr)) < 0)
 112         return ret;
 113
 114     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
 115     s->next_pts        = AV_NOPTS_VALUE;
 116
 117     av_get_channel_layout_string(buf1, sizeof(buf1),
 118                                  -1, inlink ->channel_layout);
 119     av_get_channel_layout_string(buf2, sizeof(buf2),
 120                                  -1, outlink->channel_layout);
 121     av_log(ctx, AV_LOG_VERBOSE,
 122 #if FF_API_SAMPLERATE64
 123            "fmt:%s srate:%"PRId64" cl:%s -> fmt:%s srate:%"PRId64" cl:%s\n",
 124 #else
 125            "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
 126 #endif /* FF_API_SAMPLERATE64 */
 127            av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
 128            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
 129
 130     return 0;
 131 }
 132
 133 static int request_frame(AVFilterLink *outlink)
 134 {
 135     AVFilterContext *ctx = outlink->src;
 136     ResampleContext   *s = ctx->priv;
 137     int ret = ff_request_frame(ctx->inputs[0]);
 138
 139     /* flush the lavr delay buffer */
 140     if (ret == AVERROR_EOF && s->avr) {
 141         AVFilterBufferRef *buf;
 142         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
 143                                         outlink->sample_rate,
 144                                         ctx->inputs[0]->sample_rate,
 145                                         AV_ROUND_UP);
 146
 147         if (!nb_samples)
 148             return ret;
 149
 150         buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
 151         if (!buf)
 152             return AVERROR(ENOMEM);
 153
 154         ret = avresample_convert(s->avr, (void**)buf->extended_data,
 155                                  buf->linesize[0], nb_samples,
 156                                  NULL, 0, 0);
 157         if (ret <= 0) {
 158             avfilter_unref_buffer(buf);
 159             return (ret == 0) ? AVERROR_EOF : ret;
 160         }
 161
 162         buf->pts = s->next_pts;
 163         ff_filter_samples(outlink, buf);
 164         return 0;
 165     }
 166     return ret;
 167 }
 168
 169 static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 170 {
 171     AVFilterContext  *ctx = inlink->dst;
 172     ResampleContext    *s = ctx->priv;
 173     AVFilterLink *outlink = ctx->outputs[0];
 174
 175     if (s->avr) {
 176         AVFilterBufferRef *buf_out;
 177         int delay, nb_samples, ret;
 178
 179         /* maximum possible samples lavr can output */
 180         delay      = avresample_get_delay(s->avr);
 181         nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
 182                                     outlink->sample_rate, inlink->sample_rate,
 183                                     AV_ROUND_UP);
 184
 185         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
 186         ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
 187                                      buf_out->linesize[0], nb_samples,
 188                                      (void**)buf->extended_data, buf->linesize[0],
 189                                      buf->audio->nb_samples);
 190
 191         av_assert0(!avresample_available(s->avr));
 192
 193         if (s->next_pts == AV_NOPTS_VALUE) {
 194             if (buf->pts == AV_NOPTS_VALUE) {
 195                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
 196                        "assuming 0.\n");
 197                 s->next_pts = 0;
 198             } else
 199                 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
 200                                            outlink->time_base);
 201         }
 202
 203         if (ret > 0) {
 204             buf_out->audio->nb_samples = ret;
 205             if (buf->pts != AV_NOPTS_VALUE) {
 206                 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
 207                                             outlink->time_base) -
 208                                av_rescale(delay, outlink->sample_rate,
 209                                           inlink->sample_rate);
 210             } else
 211                 buf_out->pts = s->next_pts;
 212
 213             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
 214
 215             ff_filter_samples(outlink, buf_out);
 216         }
 217         avfilter_unref_buffer(buf);
 218     } else
 219         ff_filter_samples(outlink, buf);
 220 }
 221
 222 AVFilter avfilter_af_resample = {
 223     .name          = "resample",
 224     .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
 225     .priv_size     = sizeof(ResampleContext),
 226
 227     .uninit         = uninit,
 228     .query_formats  = query_formats,
 229
 230     .inputs    = (const AVFilterPad[]) {{ .name            = "default",
 231                                           .type            = AVMEDIA_TYPE_AUDIO,
 232                                           .filter_samples  = filter_samples,
 233                                           .min_perms       = AV_PERM_READ },
 234                                         { .name = NULL}},
 235     .outputs   = (const AVFilterPad[]) {{ .name          = "default",
 236                                           .type          = AVMEDIA_TYPE_AUDIO,
 237                                           .config_props  = config_output,
 238                                           .request_frame = request_frame },
 239                                         { .name = NULL}},
 240 };