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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/audioconvert.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, const char *args)
60 {
61  VolumeContext *vol = ctx->priv;
62  static const char *shorthand[] = { "volume", "precision", NULL };
63  int ret;
64 
65  vol->class = &volume_class;
66  av_opt_set_defaults(vol);
67 
68  if ((ret = av_opt_set_from_string(vol, args, shorthand, "=", ":")) < 0)
69  return ret;
70 
71  if (vol->precision == PRECISION_FIXED) {
72  vol->volume_i = (int)(vol->volume * 256 + 0.5);
73  vol->volume = vol->volume_i / 256.0;
74  av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
75  vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
76  } else {
77  av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
78  vol->volume, 20.0*log(vol->volume)/M_LN10,
79  precision_str[vol->precision]);
80  }
81 
82  av_opt_free(vol);
83  return ret;
84 }
85 
86 static int query_formats(AVFilterContext *ctx)
87 {
88  VolumeContext *vol = ctx->priv;
89  AVFilterFormats *formats = NULL;
90  AVFilterChannelLayouts *layouts;
91  static const enum AVSampleFormat sample_fmts[][7] = {
92  /* PRECISION_FIXED */
93  {
94  AV_SAMPLE_FMT_U8,
95  AV_SAMPLE_FMT_U8P,
96  AV_SAMPLE_FMT_S16,
97  AV_SAMPLE_FMT_S16P,
98  AV_SAMPLE_FMT_S32,
99  AV_SAMPLE_FMT_S32P,
100  AV_SAMPLE_FMT_NONE
101  },
102  /* PRECISION_FLOAT */
103  {
104  AV_SAMPLE_FMT_FLT,
105  AV_SAMPLE_FMT_FLTP,
106  AV_SAMPLE_FMT_NONE
107  },
108  /* PRECISION_DOUBLE */
109  {
110  AV_SAMPLE_FMT_DBL,
111  AV_SAMPLE_FMT_DBLP,
112  AV_SAMPLE_FMT_NONE
113  }
114  };
115 
116  layouts = ff_all_channel_layouts();
117  if (!layouts)
118  return AVERROR(ENOMEM);
119  ff_set_common_channel_layouts(ctx, layouts);
120 
121  formats = ff_make_format_list(sample_fmts[vol->precision]);
122  if (!formats)
123  return AVERROR(ENOMEM);
124  ff_set_common_formats(ctx, formats);
125 
126  formats = ff_all_samplerates();
127  if (!formats)
128  return AVERROR(ENOMEM);
129  ff_set_common_samplerates(ctx, formats);
130 
131  return 0;
132 }
133 
134 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
135  int nb_samples, int volume)
136 {
137  int i;
138  for (i = 0; i < nb_samples; i++)
139  dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
140 }
141 
142 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
143  int nb_samples, int volume)
144 {
145  int i;
146  for (i = 0; i < nb_samples; i++)
147  dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
148 }
149 
150 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
151  int nb_samples, int volume)
152 {
153  int i;
154  int16_t *smp_dst = (int16_t *)dst;
155  const int16_t *smp_src = (const int16_t *)src;
156  for (i = 0; i < nb_samples; i++)
157  smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
158 }
159 
160 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
161  int nb_samples, int volume)
162 {
163  int i;
164  int16_t *smp_dst = (int16_t *)dst;
165  const int16_t *smp_src = (const int16_t *)src;
166  for (i = 0; i < nb_samples; i++)
167  smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
168 }
169 
170 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
171  int nb_samples, int volume)
172 {
173  int i;
174  int32_t *smp_dst = (int32_t *)dst;
175  const int32_t *smp_src = (const int32_t *)src;
176  for (i = 0; i < nb_samples; i++)
177  smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
178 }
179 
180 static void volume_init(VolumeContext *vol)
181 {
182  vol->samples_align = 1;
183 
184  switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
185  case AV_SAMPLE_FMT_U8:
186  if (vol->volume_i < 0x1000000)
187  vol->scale_samples = scale_samples_u8_small;
188  else
189  vol->scale_samples = scale_samples_u8;
190  break;
191  case AV_SAMPLE_FMT_S16:
192  if (vol->volume_i < 0x10000)
193  vol->scale_samples = scale_samples_s16_small;
194  else
195  vol->scale_samples = scale_samples_s16;
196  break;
197  case AV_SAMPLE_FMT_S32:
198  vol->scale_samples = scale_samples_s32;
199  break;
200  case AV_SAMPLE_FMT_FLT:
201  avpriv_float_dsp_init(&vol->fdsp, 0);
202  vol->samples_align = 4;
203  break;
204  case AV_SAMPLE_FMT_DBL:
205  avpriv_float_dsp_init(&vol->fdsp, 0);
206  vol->samples_align = 8;
207  break;
208  }
209 
210  if (ARCH_X86)
211  ff_volume_init_x86(vol);
212 }
213 
214 static int config_output(AVFilterLink *outlink)
215 {
216  AVFilterContext *ctx = outlink->src;
217  VolumeContext *vol = ctx->priv;
218  AVFilterLink *inlink = ctx->inputs[0];
219 
220  vol->sample_fmt = inlink->format;
221  vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
222  vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
223 
224  volume_init(vol);
225 
226  return 0;
227 }
228 
229 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
230 {
231  VolumeContext *vol = inlink->dst->priv;
232  AVFilterLink *outlink = inlink->dst->outputs[0];
233  int nb_samples = buf->audio->nb_samples;
234  AVFilterBufferRef *out_buf;
235 
236  if (vol->volume == 1.0 || vol->volume_i == 256)
237  return ff_filter_frame(outlink, buf);
238 
239  /* do volume scaling in-place if input buffer is writable */
240  if (buf->perms & AV_PERM_WRITE) {
241  out_buf = buf;
242  } else {
243  out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
244  if (!out_buf)
245  return AVERROR(ENOMEM);
246  out_buf->pts = buf->pts;
247  }
248 
249  if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
250  int p, plane_samples;
251 
252  if (av_sample_fmt_is_planar(buf->format))
253  plane_samples = FFALIGN(nb_samples, vol->samples_align);
254  else
255  plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
256 
257  if (vol->precision == PRECISION_FIXED) {
258  for (p = 0; p < vol->planes; p++) {
259  vol->scale_samples(out_buf->extended_data[p],
260  buf->extended_data[p], plane_samples,
261  vol->volume_i);
262  }
263  } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
264  for (p = 0; p < vol->planes; p++) {
265  vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
266  (const float *)buf->extended_data[p],
267  vol->volume, plane_samples);
268  }
269  } else {
270  for (p = 0; p < vol->planes; p++) {
271  vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
272  (const double *)buf->extended_data[p],
273  vol->volume, plane_samples);
274  }
275  }
276  }
277 
278  if (buf != out_buf)
279  avfilter_unref_buffer(buf);
280 
281  return ff_filter_frame(outlink, out_buf);
282 }
283 
284 static const AVFilterPad avfilter_af_volume_inputs[] = {
285  {
286  .name = "default",
287  .type = AVMEDIA_TYPE_AUDIO,
288  .filter_frame = filter_frame,
289  },
290  { NULL }
291 };
292 
293 static const AVFilterPad avfilter_af_volume_outputs[] = {
294  {
295  .name = "default",
296  .type = AVMEDIA_TYPE_AUDIO,
297  .config_props = config_output,
298  },
299  { NULL }
300 };
301 
302 AVFilter avfilter_af_volume = {
303  .name = "volume",
304  .description = NULL_IF_CONFIG_SMALL("Change input volume."),
305  .query_formats = query_formats,
306  .priv_size = sizeof(VolumeContext),
307  .init = init,
308  .inputs = avfilter_af_volume_inputs,
309  .outputs = avfilter_af_volume_outputs,
310  .priv_class = &volume_class,
311 };
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