2 * Copyright (c) 2011 Stefano Sabatini
3 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
5 * This file is part of FFmpeg.
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.
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.
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
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"
36 #include "af_volume.h"
38 static const char *precision_str[] = {
39 "fixed", "float", "double"
42 #define OFFSET(x) offsetof(VolumeContext, x)
43 #define A AV_OPT_FLAG_AUDIO_PARAM
44 #define F AV_OPT_FLAG_FILTERING_PARAM
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" },
57 AVFILTER_DEFINE_CLASS(volume);
59 static av_cold int init(AVFilterContext *ctx, const char *args)
61 VolumeContext *vol = ctx->priv;
62 static const char *shorthand[] = { "volume", "precision", NULL };
65 vol->class = &volume_class;
66 av_opt_set_defaults(vol);
68 if ((ret = av_opt_set_from_string(vol, args, shorthand, "=", ":")) < 0)
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);
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]);
86 static int query_formats(AVFilterContext *ctx)
88 VolumeContext *vol = ctx->priv;
89 AVFilterFormats *formats = NULL;
90 AVFilterChannelLayouts *layouts;
91 static const enum AVSampleFormat sample_fmts[][7] = {
102 /* PRECISION_FLOAT */
108 /* PRECISION_DOUBLE */
116 layouts = ff_all_channel_layouts();
118 return AVERROR(ENOMEM);
119 ff_set_common_channel_layouts(ctx, layouts);
121 formats = ff_make_format_list(sample_fmts[vol->precision]);
123 return AVERROR(ENOMEM);
124 ff_set_common_formats(ctx, formats);
126 formats = ff_all_samplerates();
128 return AVERROR(ENOMEM);
129 ff_set_common_samplerates(ctx, formats);
134 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
135 int nb_samples, int volume)
138 for (i = 0; i < nb_samples; i++)
139 dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
142 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
143 int nb_samples, int volume)
146 for (i = 0; i < nb_samples; i++)
147 dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
150 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
151 int nb_samples, int volume)
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);
160 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
161 int nb_samples, int volume)
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);
170 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
171 int nb_samples, int volume)
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));
180 static void volume_init(VolumeContext *vol)
182 vol->samples_align = 1;
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;
189 vol->scale_samples = scale_samples_u8;
191 case AV_SAMPLE_FMT_S16:
192 if (vol->volume_i < 0x10000)
193 vol->scale_samples = scale_samples_s16_small;
195 vol->scale_samples = scale_samples_s16;
197 case AV_SAMPLE_FMT_S32:
198 vol->scale_samples = scale_samples_s32;
200 case AV_SAMPLE_FMT_FLT:
201 avpriv_float_dsp_init(&vol->fdsp, 0);
202 vol->samples_align = 4;
204 case AV_SAMPLE_FMT_DBL:
205 avpriv_float_dsp_init(&vol->fdsp, 0);
206 vol->samples_align = 8;
211 ff_volume_init_x86(vol);
214 static int config_output(AVFilterLink *outlink)
216 AVFilterContext *ctx = outlink->src;
217 VolumeContext *vol = ctx->priv;
218 AVFilterLink *inlink = ctx->inputs[0];
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;
229 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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;
236 if (vol->volume == 1.0 || vol->volume_i == 256)
237 return ff_filter_frame(outlink, buf);
239 /* do volume scaling in-place if input buffer is writable */
240 if (buf->perms & AV_PERM_WRITE) {
243 out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
245 return AVERROR(ENOMEM);
246 out_buf->pts = buf->pts;
249 if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
250 int p, plane_samples;
252 if (av_sample_fmt_is_planar(buf->format))
253 plane_samples = FFALIGN(nb_samples, vol->samples_align);
255 plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
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,
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);
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);
279 avfilter_unref_buffer(buf);
281 return ff_filter_frame(outlink, out_buf);
284 static const AVFilterPad avfilter_af_volume_inputs[] = {
287 .type = AVMEDIA_TYPE_AUDIO,
288 .filter_frame = filter_frame,
293 static const AVFilterPad avfilter_af_volume_outputs[] = {
296 .type = AVMEDIA_TYPE_AUDIO,
297 .config_props = config_output,
302 AVFilter avfilter_af_volume = {
304 .description = NULL_IF_CONFIG_SMALL("Change input volume."),
305 .query_formats = query_formats,
306 .priv_size = sizeof(VolumeContext),
308 .inputs = avfilter_af_volume_inputs,
309 .outputs = avfilter_af_volume_outputs,
310 .priv_class = &volume_class,