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