h264_metadata: Add option to delete filler data
[libav.git] / libavfilter / 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 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
0651e892 27#include "libavutil/channel_layout.h"
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28#include "libavutil/common.h"
29#include "libavutil/eval.h"
30#include "libavutil/float_dsp.h"
06c3cd3c 31#include "libavutil/intreadwrite.h"
b384e031 32#include "libavutil/opt.h"
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33#include "libavutil/replaygain.h"
34
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35#include "audio.h"
36#include "avfilter.h"
37#include "formats.h"
38#include "internal.h"
39#include "af_volume.h"
40
fd9212f2 41static const char * const precision_str[] = {
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42 "fixed", "float", "double"
43};
44
45#define OFFSET(x) offsetof(VolumeContext, x)
46#define A AV_OPT_FLAG_AUDIO_PARAM
47
48static const AVOption options[] = {
49 { "volume", "Volume adjustment.",
50 OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A },
51 { "precision", "Mathematical precision.",
52 OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A, "precision" },
53 { "fixed", "8-bit fixed-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A, "precision" },
54 { "float", "32-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A, "precision" },
55 { "double", "64-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A, "precision" },
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56 { "replaygain", "Apply replaygain side data when present",
57 OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A, "replaygain" },
58 { "drop", "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP }, 0, 0, A, "replaygain" },
59 { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A, "replaygain" },
60 { "track", "track gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK }, 0, 0, A, "replaygain" },
61 { "album", "album gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM }, 0, 0, A, "replaygain" },
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62 { "replaygain_preamp", "Apply replaygain pre-amplification",
63 OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A },
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64 { "replaygain_noclip", "Apply replaygain clipping prevention",
65 OFFSET(replaygain_noclip), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, A },
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66 { NULL },
67};
68
69static const AVClass volume_class = {
70 .class_name = "volume filter",
71 .item_name = av_default_item_name,
72 .option = options,
73 .version = LIBAVUTIL_VERSION_INT,
74};
75
c43a7eca 76static av_cold int init(AVFilterContext *ctx)
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77{
78 VolumeContext *vol = ctx->priv;
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79
80 if (vol->precision == PRECISION_FIXED) {
81 vol->volume_i = (int)(vol->volume * 256 + 0.5);
82 vol->volume = vol->volume_i / 256.0;
83 av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
84 vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
85 } else {
86 av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
87 vol->volume, 20.0*log(vol->volume)/M_LN10,
88 precision_str[vol->precision]);
89 }
90
b13623e1 91 return 0;
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92}
93
94static int query_formats(AVFilterContext *ctx)
95{
96 VolumeContext *vol = ctx->priv;
97 AVFilterFormats *formats = NULL;
98 AVFilterChannelLayouts *layouts;
99 static const enum AVSampleFormat sample_fmts[][7] = {
100 /* PRECISION_FIXED */
101 {
102 AV_SAMPLE_FMT_U8,
103 AV_SAMPLE_FMT_U8P,
104 AV_SAMPLE_FMT_S16,
105 AV_SAMPLE_FMT_S16P,
106 AV_SAMPLE_FMT_S32,
107 AV_SAMPLE_FMT_S32P,
108 AV_SAMPLE_FMT_NONE
109 },
110 /* PRECISION_FLOAT */
111 {
112 AV_SAMPLE_FMT_FLT,
113 AV_SAMPLE_FMT_FLTP,
114 AV_SAMPLE_FMT_NONE
115 },
116 /* PRECISION_DOUBLE */
117 {
118 AV_SAMPLE_FMT_DBL,
119 AV_SAMPLE_FMT_DBLP,
120 AV_SAMPLE_FMT_NONE
121 }
122 };
123
124 layouts = ff_all_channel_layouts();
125 if (!layouts)
126 return AVERROR(ENOMEM);
127 ff_set_common_channel_layouts(ctx, layouts);
128
129 formats = ff_make_format_list(sample_fmts[vol->precision]);
130 if (!formats)
131 return AVERROR(ENOMEM);
132 ff_set_common_formats(ctx, formats);
133
134 formats = ff_all_samplerates();
135 if (!formats)
136 return AVERROR(ENOMEM);
137 ff_set_common_samplerates(ctx, formats);
138
139 return 0;
140}
141
142static inline void scale_samples_u8(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(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
148}
149
150static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
151 int nb_samples, int volume)
152{
153 int i;
154 for (i = 0; i < nb_samples; i++)
155 dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
156}
157
158static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
159 int nb_samples, int volume)
160{
161 int i;
162 int16_t *smp_dst = (int16_t *)dst;
163 const int16_t *smp_src = (const int16_t *)src;
164 for (i = 0; i < nb_samples; i++)
165 smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
166}
167
168static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
169 int nb_samples, int volume)
170{
171 int i;
172 int16_t *smp_dst = (int16_t *)dst;
173 const int16_t *smp_src = (const int16_t *)src;
174 for (i = 0; i < nb_samples; i++)
175 smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
176}
177
178static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
179 int nb_samples, int volume)
180{
181 int i;
182 int32_t *smp_dst = (int32_t *)dst;
183 const int32_t *smp_src = (const int32_t *)src;
184 for (i = 0; i < nb_samples; i++)
185 smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
186}
187
188
189
093804a9 190static av_cold void volume_init(VolumeContext *vol)
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191{
192 vol->samples_align = 1;
193
194 switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
195 case AV_SAMPLE_FMT_U8:
196 if (vol->volume_i < 0x1000000)
197 vol->scale_samples = scale_samples_u8_small;
198 else
199 vol->scale_samples = scale_samples_u8;
200 break;
201 case AV_SAMPLE_FMT_S16:
202 if (vol->volume_i < 0x10000)
203 vol->scale_samples = scale_samples_s16_small;
204 else
205 vol->scale_samples = scale_samples_s16;
206 break;
207 case AV_SAMPLE_FMT_S32:
208 vol->scale_samples = scale_samples_s32;
209 break;
210 case AV_SAMPLE_FMT_FLT:
211 avpriv_float_dsp_init(&vol->fdsp, 0);
212 vol->samples_align = 4;
213 break;
214 case AV_SAMPLE_FMT_DBL:
215 avpriv_float_dsp_init(&vol->fdsp, 0);
216 vol->samples_align = 8;
217 break;
218 }
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219
220 if (ARCH_X86)
221 ff_volume_init_x86(vol);
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222}
223
224static int config_output(AVFilterLink *outlink)
225{
226 AVFilterContext *ctx = outlink->src;
227 VolumeContext *vol = ctx->priv;
228 AVFilterLink *inlink = ctx->inputs[0];
229
230 vol->sample_fmt = inlink->format;
231 vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
232 vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
233
234 volume_init(vol);
235
236 return 0;
237}
238
7e350379 239static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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240{
241 VolumeContext *vol = inlink->dst->priv;
242 AVFilterLink *outlink = inlink->dst->outputs[0];
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243 int nb_samples = buf->nb_samples;
244 AVFrame *out_buf;
06c3cd3c 245 AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
39c2880e 246 int ret;
b384e031 247
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248 if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
249 if (vol->replaygain != REPLAYGAIN_DROP) {
250 AVReplayGain *replaygain = (AVReplayGain*)sd->data;
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251 int32_t gain = 100000;
252 uint32_t peak = 100000;
253 float g, p;
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254
255 if (vol->replaygain == REPLAYGAIN_TRACK &&
aaab192d 256 replaygain->track_gain != INT32_MIN) {
06c3cd3c 257 gain = replaygain->track_gain;
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258
259 if (replaygain->track_peak != 0)
260 peak = replaygain->track_peak;
261 } else if (replaygain->album_gain != INT32_MIN) {
06c3cd3c 262 gain = replaygain->album_gain;
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263
264 if (replaygain->album_peak != 0)
265 peak = replaygain->album_peak;
266 } else {
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267 av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
268 "values are unknown.\n");
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269 }
270 g = gain / 100000.0f;
aaab192d 271 p = peak / 100000.0f;
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272
273 av_log(inlink->dst, AV_LOG_VERBOSE,
274 "Using gain %f dB from replaygain side data.\n", g);
275
a49aa440 276 vol->volume = pow(10, (g + vol->replaygain_preamp) / 20);
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277 if (vol->replaygain_noclip)
278 vol->volume = FFMIN(vol->volume, 1.0 / p);
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279 vol->volume_i = (int)(vol->volume * 256 + 0.5);
280
281 volume_init(vol);
282 }
283 av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
284 }
285
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286 if (vol->volume == 1.0 || vol->volume_i == 256)
287 return ff_filter_frame(outlink, buf);
288
289 /* do volume scaling in-place if input buffer is writable */
7e350379 290 if (av_frame_is_writable(buf)) {
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291 out_buf = buf;
292 } else {
7e350379 293 out_buf = ff_get_audio_buffer(inlink, nb_samples);
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294 if (!out_buf)
295 return AVERROR(ENOMEM);
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296 ret = av_frame_copy_props(out_buf, buf);
297 if (ret < 0) {
298 av_frame_free(&out_buf);
299 av_frame_free(&buf);
300 return ret;
301 }
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302 }
303
304 if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
305 int p, plane_samples;
306
307 if (av_sample_fmt_is_planar(buf->format))
308 plane_samples = FFALIGN(nb_samples, vol->samples_align);
309 else
310 plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
311
312 if (vol->precision == PRECISION_FIXED) {
313 for (p = 0; p < vol->planes; p++) {
314 vol->scale_samples(out_buf->extended_data[p],
315 buf->extended_data[p], plane_samples,
316 vol->volume_i);
317 }
318 } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
319 for (p = 0; p < vol->planes; p++) {
320 vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
321 (const float *)buf->extended_data[p],
322 vol->volume, plane_samples);
323 }
324 } else {
325 for (p = 0; p < vol->planes; p++) {
326 vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
327 (const double *)buf->extended_data[p],
328 vol->volume, plane_samples);
329 }
330 }
331 }
332
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333 emms_c();
334
b384e031 335 if (buf != out_buf)
7e350379 336 av_frame_free(&buf);
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337
338 return ff_filter_frame(outlink, out_buf);
339}
340
341static const AVFilterPad avfilter_af_volume_inputs[] = {
342 {
343 .name = "default",
344 .type = AVMEDIA_TYPE_AUDIO,
345 .filter_frame = filter_frame,
346 },
347 { NULL }
348};
349
350static const AVFilterPad avfilter_af_volume_outputs[] = {
351 {
352 .name = "default",
353 .type = AVMEDIA_TYPE_AUDIO,
354 .config_props = config_output,
355 },
356 { NULL }
357};
358
cd43ca04 359AVFilter ff_af_volume = {
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360 .name = "volume",
361 .description = NULL_IF_CONFIG_SMALL("Change input volume."),
362 .query_formats = query_formats,
363 .priv_size = sizeof(VolumeContext),
b13623e1 364 .priv_class = &volume_class,
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365 .init = init,
366 .inputs = avfilter_af_volume_inputs,
367 .outputs = avfilter_af_volume_outputs,
368};