libavfilter/af_volume.c

   1 /*
   2  * Copyright (c) 2011 Stefano Sabatini
   3  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
   4  *
   5  * This file is part of Libav.
   6  *
   7  * Libav 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  * Libav 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 Libav; 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  * audio volume filter
  25  */
  26
  27 #include "libavutil/channel_layout.h"
  28 #include "libavutil/common.h"
  29 #include "libavutil/eval.h"
  30 #include "libavutil/float_dsp.h"
  31 #include "libavutil/opt.h"
  32 #include "audio.h"
  33 #include "avfilter.h"
  34 #include "formats.h"
  35 #include "internal.h"
  36 #include "af_volume.h"
  37
  38 static const char *precision_str[] = {
  39     "fixed", "float", "double"
  40 };
  41
  42 #define OFFSET(x) offsetof(VolumeContext, x)
  43 #define A AV_OPT_FLAG_AUDIO_PARAM
  44
  45 static const AVOption options[] = {
  46     { "volume", "Volume adjustment.",
  47             OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A },
  48     { "precision", "Mathematical precision.",
  49             OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A, "precision" },
  50         { "fixed",  "8-bit fixed-point.",     0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED  }, INT_MIN, INT_MAX, A, "precision" },
  51         { "float",  "32-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT  }, INT_MIN, INT_MAX, A, "precision" },
  52         { "double", "64-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A, "precision" },
  53     { NULL },
  54 };
  55
  56 static const AVClass volume_class = {
  57     .class_name = "volume filter",
  58     .item_name  = av_default_item_name,
  59     .option     = options,
  60     .version    = LIBAVUTIL_VERSION_INT,
  61 };
  62
  63 static av_cold int init(AVFilterContext *ctx)
  64 {
  65     VolumeContext *vol = ctx->priv;
  66
  67     if (vol->precision == PRECISION_FIXED) {
  68         vol->volume_i = (int)(vol->volume * 256 + 0.5);
  69         vol->volume   = vol->volume_i / 256.0;
  70         av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
  71                vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
  72     } else {
  73         av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
  74                vol->volume, 20.0*log(vol->volume)/M_LN10,
  75                precision_str[vol->precision]);
  76     }
  77
  78     return 0;
  79 }
  80
  81 static int query_formats(AVFilterContext *ctx)
  82 {
  83     VolumeContext *vol = ctx->priv;
  84     AVFilterFormats *formats = NULL;
  85     AVFilterChannelLayouts *layouts;
  86     static const enum AVSampleFormat sample_fmts[][7] = {
  87         /* PRECISION_FIXED */
  88         {
  89             AV_SAMPLE_FMT_U8,
  90             AV_SAMPLE_FMT_U8P,
  91             AV_SAMPLE_FMT_S16,
  92             AV_SAMPLE_FMT_S16P,
  93             AV_SAMPLE_FMT_S32,
  94             AV_SAMPLE_FMT_S32P,
  95             AV_SAMPLE_FMT_NONE
  96         },
  97         /* PRECISION_FLOAT */
  98         {
  99             AV_SAMPLE_FMT_FLT,
 100             AV_SAMPLE_FMT_FLTP,
 101             AV_SAMPLE_FMT_NONE
 102         },
 103         /* PRECISION_DOUBLE */
 104         {
 105             AV_SAMPLE_FMT_DBL,
 106             AV_SAMPLE_FMT_DBLP,
 107             AV_SAMPLE_FMT_NONE
 108         }
 109     };
 110
 111     layouts = ff_all_channel_layouts();
 112     if (!layouts)
 113         return AVERROR(ENOMEM);
 114     ff_set_common_channel_layouts(ctx, layouts);
 115
 116     formats = ff_make_format_list(sample_fmts[vol->precision]);
 117     if (!formats)
 118         return AVERROR(ENOMEM);
 119     ff_set_common_formats(ctx, formats);
 120
 121     formats = ff_all_samplerates();
 122     if (!formats)
 123         return AVERROR(ENOMEM);
 124     ff_set_common_samplerates(ctx, formats);
 125
 126     return 0;
 127 }
 128
 129 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
 130                                     int nb_samples, int volume)
 131 {
 132     int i;
 133     for (i = 0; i < nb_samples; i++)
 134         dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
 135 }
 136
 137 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
 138                                           int nb_samples, int volume)
 139 {
 140     int i;
 141     for (i = 0; i < nb_samples; i++)
 142         dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
 143 }
 144
 145 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
 146                                      int nb_samples, int volume)
 147 {
 148     int i;
 149     int16_t *smp_dst       = (int16_t *)dst;
 150     const int16_t *smp_src = (const int16_t *)src;
 151     for (i = 0; i < nb_samples; i++)
 152         smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
 153 }
 154
 155 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
 156                                            int nb_samples, int volume)
 157 {
 158     int i;
 159     int16_t *smp_dst       = (int16_t *)dst;
 160     const int16_t *smp_src = (const int16_t *)src;
 161     for (i = 0; i < nb_samples; i++)
 162         smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
 163 }
 164
 165 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
 166                                      int nb_samples, int volume)
 167 {
 168     int i;
 169     int32_t *smp_dst       = (int32_t *)dst;
 170     const int32_t *smp_src = (const int32_t *)src;
 171     for (i = 0; i < nb_samples; i++)
 172         smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
 173 }
 174
 175
 176
 177 static av_cold void volume_init(VolumeContext *vol)
 178 {
 179     vol->samples_align = 1;
 180
 181     switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
 182     case AV_SAMPLE_FMT_U8:
 183         if (vol->volume_i < 0x1000000)
 184             vol->scale_samples = scale_samples_u8_small;
 185         else
 186             vol->scale_samples = scale_samples_u8;
 187         break;
 188     case AV_SAMPLE_FMT_S16:
 189         if (vol->volume_i < 0x10000)
 190             vol->scale_samples = scale_samples_s16_small;
 191         else
 192             vol->scale_samples = scale_samples_s16;
 193         break;
 194     case AV_SAMPLE_FMT_S32:
 195         vol->scale_samples = scale_samples_s32;
 196         break;
 197     case AV_SAMPLE_FMT_FLT:
 198         avpriv_float_dsp_init(&vol->fdsp, 0);
 199         vol->samples_align = 4;
 200         break;
 201     case AV_SAMPLE_FMT_DBL:
 202         avpriv_float_dsp_init(&vol->fdsp, 0);
 203         vol->samples_align = 8;
 204         break;
 205     }
 206
 207     if (ARCH_X86)
 208         ff_volume_init_x86(vol);
 209 }
 210
 211 static int config_output(AVFilterLink *outlink)
 212 {
 213     AVFilterContext *ctx = outlink->src;
 214     VolumeContext *vol   = ctx->priv;
 215     AVFilterLink *inlink = ctx->inputs[0];
 216
 217     vol->sample_fmt = inlink->format;
 218     vol->channels   = av_get_channel_layout_nb_channels(inlink->channel_layout);
 219     vol->planes     = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
 220
 221     volume_init(vol);
 222
 223     return 0;
 224 }
 225
 226 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
 227 {
 228     VolumeContext *vol    = inlink->dst->priv;
 229     AVFilterLink *outlink = inlink->dst->outputs[0];
 230     int nb_samples        = buf->nb_samples;
 231     AVFrame *out_buf;
 232
 233     if (vol->volume == 1.0 || vol->volume_i == 256)
 234         return ff_filter_frame(outlink, buf);
 235
 236     /* do volume scaling in-place if input buffer is writable */
 237     if (av_frame_is_writable(buf)) {
 238         out_buf = buf;
 239     } else {
 240         out_buf = ff_get_audio_buffer(inlink, nb_samples);
 241         if (!out_buf)
 242             return AVERROR(ENOMEM);
 243         out_buf->pts = buf->pts;
 244     }
 245
 246     if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
 247         int p, plane_samples;
 248
 249         if (av_sample_fmt_is_planar(buf->format))
 250             plane_samples = FFALIGN(nb_samples, vol->samples_align);
 251         else
 252             plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
 253
 254         if (vol->precision == PRECISION_FIXED) {
 255             for (p = 0; p < vol->planes; p++) {
 256                 vol->scale_samples(out_buf->extended_data[p],
 257                                    buf->extended_data[p], plane_samples,
 258                                    vol->volume_i);
 259             }
 260         } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
 261             for (p = 0; p < vol->planes; p++) {
 262                 vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
 263                                              (const float *)buf->extended_data[p],
 264                                              vol->volume, plane_samples);
 265             }
 266         } else {
 267             for (p = 0; p < vol->planes; p++) {
 268                 vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
 269                                              (const double *)buf->extended_data[p],
 270                                              vol->volume, plane_samples);
 271             }
 272         }
 273     }
 274
 275     if (buf != out_buf)
 276         av_frame_free(&buf);
 277
 278     return ff_filter_frame(outlink, out_buf);
 279 }
 280
 281 static const AVFilterPad avfilter_af_volume_inputs[] = {
 282     {
 283         .name           = "default",
 284         .type           = AVMEDIA_TYPE_AUDIO,
 285         .filter_frame   = filter_frame,
 286     },
 287     { NULL }
 288 };
 289
 290 static const AVFilterPad avfilter_af_volume_outputs[] = {
 291     {
 292         .name         = "default",
 293         .type         = AVMEDIA_TYPE_AUDIO,
 294         .config_props = config_output,
 295     },
 296     { NULL }
 297 };
 298
 299 AVFilter ff_af_volume = {
 300     .name           = "volume",
 301     .description    = NULL_IF_CONFIG_SMALL("Change input volume."),
 302     .query_formats  = query_formats,
 303     .priv_size      = sizeof(VolumeContext),
 304     .priv_class     = &volume_class,
 305     .init           = init,
 306     .inputs         = avfilter_af_volume_inputs,
 307     .outputs        = avfilter_af_volume_outputs,
 308 };